- sr-users - kamailio.org

[Serusers] ISDN Screening indicator
by Razvan Nemesiu 14 Jun '05

14 Jun '05

Hi all, Does anyone know how to modify from a Cisco As5300 or 5350 the screening indicator parameter ? (its value is 1 and I would like to change it with 3-Network Provided). Thanks -- Using Opera's revolutionary e-mail client: http://www.opera.com/m2/

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[Serusers] serweb add user
by Olivier Roomans 14 Jun '05

14 Jun '05

Hi, I would like to ask a small question about serweb : how can I use it to add new users ? I've done some research on the ml archive but didn't find clear information about that (some people where talking about sending an email but I'm not really sure). Thanks in advance.

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[Serusers] Use of fifo for registration :ul_add: command tutorial
by Rodrigue ESTEVE 14 Jun '05

14 Jun '05

Hi all, I would like to use ser_fifo to register a user; I think the :ul_add: command is the clue, but I don't how to use the fifo to do this. My aim is to do the following: I have a Voip gateway on a customer with a lot of phone number (700) and I would like to authenticate each number. Unfortunately I have to add a small amount of configuration on the voip gateway each time I add a number (uri), moreover I'm not sure this gateway will work with so many numbers, and the bandwidth needed to register each number every 5 minutes will be wasted. The idea is to challenge only one time the gateway and then in ser.cfg if the challenge succeeds I execute a script to register by bulk all the numbers enclosed in a file on the proxy. To do this I have to pass arguments to the fifo using the :ul_add: command, but this is where I'm stuck. I've done this: echo :ul_add:receiving_fifo > /tmp/ser_fifo echo location > /tmp/ser_fifo echo 100 > /tmp/ser_fifo echo <sip:100@10.1.1.1:5060> > /tmp/ser_fifo echo 300 > /tmp/ser_fifo echo 0 > /tmp/ser_fifo echo 2 > /tmp/ser_fifo but it does not work, I think this is due to the receiving_fifo, I have to do something with it to process the response of SER, but what ?? I know I can use the following command to workaround the pbm: Serctl fifo ul_add location ... But it requires one process each time I register a number, and it takes 1 minute on a Celeron 700 for 500 numbers (30 % CPU), and I think using the fifo is a more suitable solution (less CPU consuming). Thanks for your support, Rodrigue PS: I would like to register every 5minutes,cause the customer gateway is able to redirect to PSTN in case of failure.

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[Serusers] Re: problem in ser
by Kamran Ahmad 14 Jun '05

14 Jun '05

i am using kernel-2.6.9-1.667 >hello > >i have some problems i am using ser-0.9.0 with >radiusclient-ng. but unable to compile radius modules. >and when i am trying to start ser. and unable to stop >ser. or even i cannot kill ser. any comments > > > >[root@voice ser-0.9.0]# serctl moni >Error opening ser's FIFO /tmp/ser_fifo >Make sure you have line fifo=/tmp/ser_fifo in your >config >----------------------------------------------------- >#ser.cfg >dns=no >rev_dns=no >fifo="/tmp/ser_fifo" > >loadmodule "/usr/local/lib/ser/modules/mysql.so" >loadmodule "/usr/local/lib/ser/modules/sl.so" >loadmodule "/usr/local/lib/ser/modules/tm.so" >loadmodule "/usr/local/lib/ser/modules/rr.so" >loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >loadmodule "/usr/local/lib/ser/modules/usrloc.so" >loadmodule "/usr/local/lib/ser/modules/registrar.so" >loadmodule "/usr/local/lib/ser/modules/xlog.so" > >loadmodule "/usr/local/lib/ser/modules/auth.so" > >loadmodule "/usr/local/lib/ser/modules/exec.so" > >loadmodule "/usr/local/lib/ser/modules/uri.so" >loadmodule "/usr/local/lib/ser/modules/nathelper.so" >loadmodule "/usr/local/lib/ser/modules/textops.so" >loadmodule "/usr/local/lib/ser/modules/auth_db.so" > >modparam("nathelper", "natping_interval", 30) >modparam("nathelper", "ping_nated_only", 1) >modparam("nathelper", >"rtpproxy_sock","unix:/var/run/rtpproxy.sock") >modparam("usrloc", "db_mode", 0) >modparam("registrar", "nat_flag", 6) >modparam("rr", "enable_full_lr", 1) >modparam("nathelper", "natping_interval", 15) >modparam("nathelper", "ping_nated_only", 1) >----------------------------------------------------- > >make_and_install_output.txt > >make[1]: Entering directory >`/ser-0.9.0/modules/auth_radius' >../../Makefile.rules:77: authorize.d: No such file or >directory >../../Makefile.rules:77: authrad_mod.d: No such file >or directory >../../Makefile.rules:77: sterman.d: No such file or >directory >make[1]: Leaving directory >`/ser-0.9.0/modules/auth_radius' >make[1]: Entering directory >`/ser-0.9.0/modules/auth_radius' >gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align >-Wall -minline-all-stringops -malign-double >-falign-loops -mtune=athlon >-I/opt/axxany/siprouter/radiusclient/include >-DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' >-DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 >-D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK >-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >-DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT >-DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 >-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD >-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL >-DHAVE_ALLOCA_H -I/usr/local/include -c authorize.c -o >authorize.o >gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align >-Wall -minline-all-stringops -malign-double >-falign-loops -mtune=athlon >-I/opt/axxany/siprouter/radiusclient/include >-DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' >-DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 >-D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK >-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >-DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT >-DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 >-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD >-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL >-DHAVE_ALLOCA_H -I/usr/local/include -c authrad_mod.c >-o authrad_mod.o >authrad_mod.c: In function `mod_init': >authrad_mod.c:110: error: `DICT_VENDOR' undeclared >(first use in this function) >authrad_mod.c:110: error: (Each undeclared identifier >is reported only once >authrad_mod.c:110: error: for each function it appears >in.) >authrad_mod.c:110: error: `vend' undeclared (first use >in this function) >authrad_mod.c:134: warning: assignment makes pointer >from integer without a cast >authrad_mod.c:139: error: too many arguments to >function `rc_conf_str' >authrad_mod.c:139: error: too many arguments to >function `rc_read_dictionary' >authrad_mod.c:144: warning: implicit declaration of >function `rc_dict_findvend' >authrad_mod.c:159: error: too many arguments to >function `rc_dict_findattr' >authrad_mod.c:159: error: too many arguments to >function `rc_dict_findval' >make[1]: *** [authrad_mod.o] Error 1 >make[1]: Leaving directory >`/ser-0.9.0/modules/auth_radius' > > > >make[1]: Entering directory >`/ser-0.9.0/modules/group_radius' >../../Makefile.rules:77: group.d: No such file or >directory >../../Makefile.rules:77: grouprad_mod.d: No such file >or directory >make[1]: Leaving directory >`/ser-0.9.0/modules/group_radius' >make[1]: Entering directory >`/ser-0.9.0/modules/group_radius' >gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align >-Wall -minline-all-stringops -malign-double >-falign-loops -mtune=athlon >-I/opt/axxany/siprouter/radiusclient/include >-DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' >-DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 >-D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' >-DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK >-DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >-DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT >-DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 >-DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD >-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL >-DHAVE_ALLOCA_H -I/usr/local/include -c group.c -o >group.o >group.c: In function `radius_is_user_in': >group.c:194: warning: passing arg 2 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:194: warning: passing arg 3 of `rc_avpair_add' >makes pointer from integer without a cast >group.c:194: warning: passing arg 4 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:194: error: too many arguments to function >`rc_avpair_add' >group.c:203: warning: passing arg 2 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:203: warning: passing arg 3 of `rc_avpair_add' >makes pointer from integer without a cast >group.c:203: warning: passing arg 4 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:203: error: too many arguments to function >`rc_avpair_add' >group.c:209: warning: passing arg 2 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:209: warning: passing arg 3 of `rc_avpair_add' >makes pointer from integer without a cast >group.c:209: warning: passing arg 4 of `rc_avpair_add' >makes integer from pointer without a cast >group.c:209: error: too many arguments to function >`rc_avpair_add' >group.c:215: warning: passing arg 1 of `rc_auth' makes >integer from pointer without a cast>group.c:215: warning: passing arg 3 of `rc_auth' from >incompatible pointer type >group.c:215: warning: passing arg 4 of `rc_auth' from >incompatible pointer type >group.c:215: error: too many arguments to function >`rc_auth' >make[1]: *** [group.o] Error 1 >make[1]: Leaving directory >`/ser-0.9.0/modules/group_radius' __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

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[Serusers] pa cannot work as a B2BUA?
by mma 14 Jun '05

14 Jun '05

hi, i read in pa module that it cannot subsribe to presence of users using SIP, so how can i do to manage the presence of my contact list i added pa module to SER and a had this errors in response to a SUBSCRIBE request parse_hfs(): Error while parsing headers handle_subscription(): Error while parsing message header please i need help regards

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[Serusers] 'subst' on multiple lines - wackiness (and possible bug ensues?
by Kanakatti Mahesh Subramanya 14 Jun '05

14 Jun '05

I'm using 'subst' in multiline mode - basically using it to rewrite a 'Refer-To' header based on the contents of the 'Contact' header. However, if I do that, it looks like all the default processing that occurs on the message (specifically, the addition of a 'Via' header), gets tacked on to the end of the message. It almost looks like all the changes that were made to the message end up in a buffer that gets appended to the end of the message, and *not* as part of the message. In the example below, I just used the following to generate the error condition subst('/(.*)5100(.*)/\15100aptela\2/si'); And yes, I know that the above doesnt need the 's' flag. Its just that i've distilled my ridiculously complex 'subst' expressiond down to the above really simple expression which still causes the error below. So, the question is, is this actually supposed to work? Anyone ever actually tried this out? e.g. incoming packet is (note that th "Refer-To" line sez 'sip:5100@...'. This will get changed to sip:5100aptela@... REFER sip:7732206484@66.23.129.219:5090 SIP/2.0. f: <sip:operator.aptela@voice.aptela.com>;tag=7f1ca8c0-13c442ad79d2. t: "7732206484"<sip:7732206484@66.23.129.219:5090>;tag=as10a69cf8. Call-ID: 6bfbb1d627b2c0366bb2be7206177cea(a)66.23.129.219. CSeq: 2 REFER. Via: SIP/2.0/UDP 192.168.28.127:5060;rport;branch=z9hG4bK-42ad79d6-1750412-5e02. Max-Forwards: 70. Supported: 100rel,replaces. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. User-Agent: SIP Phone-1.0.41 - ABP. Contact: <sip:operator.aptela@64.81.147.138:33049>. Route: <sip:69.25.47.134;lr;ftag=as10a69cf8>. Referred-By: <sip:operator.aptela@voice.aptela.com>. Refer-To: <sip:5100@voice.aptela.com>. Content-Length: 0. . outgoing packet *should* be (note the Refer-To line now REFER sip:7732206484@66.23.129.219:5090 SIP/2.0. Record-Route: <sip:69.25.47.134;ftag=7f1ca8c0-13c442ad79d2;lr=on>. f: <sip:operator.aptela@voice.aptela.com>;tag=7f1ca8c0-13c442ad79d2. t: "7732206484"<sip:7732206484@66.23.129.219:5090>;tag=as10a69cf8. Call-ID: 6bfbb1d627b2c0366bb2be7206177cea(a)66.23.129.219. CSeq: 2 REFER. Via: SIP/2.0/UDP 69.25.47.134;branch=z9hG4bKfd4a.00e82c93.0. Via: SIP/2.0/UDP 192.168.28.127:5060;received=64.81.147.138;rport=33049;branch=z9hG4bK-42ad79d6-1750412-5e02. Max-Forwards: 16. Supported: 100rel,replaces. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. User-Agent: SIP Phone-1.0.41 - ABP. Contact: <sip:operator.aptela@64.81.147.138:33049>. Route: <sip:69.25.47.134;lr;ftag=as10a69cf8>. Referred-By: <sip:operator.aptela@voice.aptela.com>. Refer-To: <sip:5100aptela@voice.aptela.com>. Content-Length: 0. but, instead, is REFER sip:7732206484@66.23.129.219:5090 SIP/2.0. Record-Route: <sip:69.25.47.134;ftag=7f1ca8c0-13c442ad79d2;lr=on>. f: <sip:operator.aptela@voice.aptela.com>;tag=7f1ca8c0-13c442ad79d2. t: "7732206484"<sip:7732206484@66.23.129.219:5090>;tag=as10a69cf8. Call-ID: 6bfbb1d627b2c0366bb2be7206177cea(a)66.23.129.219. CSeq: 2 REFER. Via: SIP/2.0/UDP 192.168.28.127:5060;rport;branch=z9hG4bK-42ad79d6-1750412-5e02. Max-Forwards: 16. Supported: 100rel,replaces. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. User-Agent: SIP Phone-1.0.41 - ABP. Contact: <sip:operator.aptela@64.81.147.138:33049>. Route: <sip:69.25.47.134;lr;ftag=as10a69cf8>. Referred-By: <sip:operator.aptela@voice.aptela.com>. Refer-To: <sip:5100aptela@voice.aptela.com>. Content-Length: 0. . Via: SIP/2.0/UDP 69.25.47.134;branch=z9hG4bKfd4a.00e82c93.0. ;received=64.81.147.138;rport=33049 note the two lines at the bottom of the message. These are the additional lines that got changed in the message as it was processed.

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[Serusers] can't establish calls whenever using outbound proxy
by jeff kwong 14 Jun '05

14 Jun '05

Hi again! I used the nathelper.cfg from the cvs to use ser as an outbound proxy with rtpproxy. I commented out some parts which I thought I would not be needing. rtpproxy is working fine as well as ser. I placed my outbound proxy with a public ip and two ipphones behind nat. The phones were able to register, however whenever I call one using the other ipphone, my request timeouts and I get a busy signal. However, when i place one ipphone to the public and the other still behind nat. I was able to recieve a ring, calling from the nated to the public. And when both phones are in public, thus bypassing the outbond proxy, my calls go perfectly through. When i checked the dumps at the outbound proxy, it seems the soft switch does not return an "ok signal " or anything at all after recieving an invite with authentication, that is whenever using an OP. Are these symptoms familiar to anybody here? Any help would be most welcome. Btw I posted my ser.cfg file here just in case. # # $Id: nathelper.cfg,v 1.1 2003/11/10 14:15:36 janakj Exp $ # # simple quick-start config script including nathelper support # This default script includes nathelper support. To make it work # you will also have to install Maxim's RTP proxy. The proxy is enforced # if one of the parties is behind a NAT. # # If you have an endpoing in the public internet which is known to # support symmetric RTP (Cisco PSTN gateway or voicemail, for example), # then you don't have to force RTP proxy. If you don't want to enforce # RTP proxy for some destinations than simply use t_relay() instead of # route(1) # # Sections marked with !! Nathelper contain modifications for nathelper # # NOTE !! This config is EXPERIMENTAL ! # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # !! Nathelper loadmodule "/usr/local/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; }

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[Serusers] Out-of-office-reply: Serusers Digest, Vol 26, Issue 20
by David.Monahan＠aepona.com 14 Jun '05

14 Jun '05

I will be out of the office until Monday 27th June. For urgent queries please contact Judith Russell (judith.russell(a)aepona.com).

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[Serusers] Error login to iptel.org SERWEB
by SS Ng 14 Jun '05

14 Jun '05

While login into http://iptel.org/user, I got the following error Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /var/www/serweb_bat/html/functions.php on line 199 Thanks ssng " name=Author> all.clsMenuItemNS, .clsMenuItemIE{text-decoration: none; font: bold 10px Verdana; color: black; cursor: hand; z-index:100} #MainTable A:hover {color: white;} var keepstatic=1 //specify whether menu should stay static (works only in IE4+) var menucolor="#75C5F0" //specify menu color var submenuwidth=150 //specify sub menus color --------------------------------- Meet your soulmate! Yahoo! Asia presents Meetic - where millions of singles gather

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[Serusers] Problem with UAC modules
by Mohammad Khan 14 Jun '05

14 Jun '05

I am using ser downloaded from cvs (cvs co -r HEAD sip_router, download on June 08, 2005 12:30 PM EST) I used the patch file found in modules/uac/doc/replace_from.patch. (Did I do right?) I am getting below, while trying to start ser Jun 8 18:37:18 sip ser: set_mod_param_regex: parameter <db_url> not found in module <acc> Jun 8 18:37:18 sip ser: parse error (49,20-21): Can't set module parameter Jun 8 18:37:18 sip ser: set_mod_param_regex: parameter <db_flag> not found in module <acc> Jun 8 18:37:18 sip ser: parse error (86,19-20): Can't set module parameter Jun 8 18:37:18 sip ser: set_mod_param_regex: parameter <db_missed_flag> not found in module <acc> Jun 8 18:37:18 sip ser: parse error (87,19-20): Can't set module parameter Jun 8 18:37:18 sip ser: ERROR: bad config file (3 errors) After uncomment acc module (just to see if it works without acc) I am getting these: Jun 8 18:50:48 sip /usr/sbin/ser[3076]: ERROR:uac:get_autenticate_hdr: reply has no auth hdr (Proxy-Authenticate) Jun 8 18:50:48 sip /usr/sbin/ser[3076]: ERROR:uac:uac_auth: failed to extract authenticate hdr Jun 8 18:50:48 sip /usr/sbin/ser[3074]: Warning: sl_send_reply: I won't send a reply for ACK!! Jun 8 18:50:49 sip /usr/sbin/ser[3076]: Warning: sl_send_reply: I won't send a reply for ACK!! Jun 8 18:50:51 sip /usr/sbin/ser[3075]: Warning: sl_send_reply: I won't send a reply for ACK!! Please help. Thanks, MOhammad

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[Serusers] Re: Problem with cancel a invite
by michael＠polylink.net 14 Jun '05

14 Jun '05

I think there was something wrong with Bs gateway other than SER. The sip singaling is ok because A received the right message from B. >Subject: [Serusers] Problem with cancel a invite >To: serusers(a)lists.iptel.org >Message-ID: <42AD5451.1000300(a)borderlight.net> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hi all, >I´ve got a problem when trying to cancel an invite. >The situation: A makes a call to B (INVITE). B press the "hang up" >button. An here comes the problem. > From As side, everything is OK, but from Bs side, the phone is still >ringing... >How can a solve this problem? Anyone? > >Thanks, >Anders Michael Zhang

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[Serusers] failed to lookup for few seconds for every one hour
by Karun Chemudugunta 14 Jun '05

14 Jun '05

Hi Group, I am seeing issues where lookup() failing for every one hour for period of few seconds. ( ~ 20 to 30 seconds). Is it known issues ? Any house keeping happening for every hour ??? Version SER software: 0.8.12 Please find my attached ser.cfg Thanks in advance. Karun _______________________________________________________________________ Jun 13 12:41:14 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 12:41:14 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request from config Jun 13 13:40:44 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 13:41:15 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 13:41:15 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 13:41:17 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 13:41:18 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4805]: lookup failed :SIP request from config Jun 13 13:41:19 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 13:41:20 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 14:40:57 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 14:40:58 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 14:40:58 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 14:40:59 encasip001 /usr/sbin/ser[4811]: lookup failed :SIP request from config Jun 13 14:41:00 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 15:40:35 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:35 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4807]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4806]: lookup failed :SIP request from config Jun 13 15:40:36 encasip001 /usr/sbin/ser[4803]: lookup failed :SIP request from config Jun 13 15:40:37 encasip001 /usr/sbin/ser[4809]: lookup failed :SIP request from config Jun 13 15:40:37 encasip001 /usr/sbin/ser[4808]: lookup failed :SIP request from config Jun 13 15:40:39 encasip001 /usr/sbin/ser[4810]: lookup failed :SIP request from config ___________________________________________________________________________ # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if(method=="OPTIONS") { sl_send_reply("200", "Ok"); break; }; if (method=="REGISTER") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location","encamed")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # setflag(1); # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { revert_uri(); if (!lookup("location","encamed")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_on_failure("6"); t_relay(); } failure_route[6] { log(1,"Inside failure_route six"); exec_msg("printenv|grep '^SIP'|mail -v -s onfailure6 krcgunta(a)yahoo.com"); }

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RE: [Serusers] using web page of mediaproxy
by Lucas Aimaretto 13 Jun '05

13 Jun '05

> > Hi all, > > > > I've used the webpage media_sessions.phtml for monitoring > > usage of mediaproxy. But it only works if the web server is > > set up at the same PC where mediaproxy is working. Is there > > any possible way of installing the web page at another PC ? > > > yes you should edit the script adding the server's host. > > Rosario Hi there, Well, actually, I did that. I replaced the IP of my mediaproxy at the file media_sessions.phtml. Look ... // List with MediaProxy servers to display sessions for. // Servers are specified in the form "IP:Port" (port can be missing) // If port is missing it is considered to be 25060 $servers = array("MY_IP:25060"); ... but it does not show the rtp sessions. However, if I run the web page at a local web server, it does show the rtp sessions. Any ideas? I bet is some web server configuration, but don’t know what. This is what I did. Alias /mediaproxy/ /home/mediaproxy/ <Directory "/home/mediaproxy/"> Options None AllowOverride AuthConfig #Order allow,deny #Order allow Allow from all </Directory> Best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 11/06/2005

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[Serusers] using web page of mediaproxy
by Lucas Aimaretto 13 Jun '05

13 Jun '05

Hi all, I've used the webpage media_sessions.phtml for monitoring usage of mediaproxy. But it only works if the web server is set up at the same PC where mediaproxy is working. Is there any possible way of installing the web page at another PC ? Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 11/06/2005

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[Serusers] Race condition with CANCEL?
by Ingo Wolfsberger 13 Jun '05

13 Jun '05

Hello, Following situation: t+0: User A is registerd User B is not registered t+1: User A calls user B Call is transfered to the vm box. vm does not go off-hook. t+2: User B registers t+3: User A hooks on, a CANCEL is send. Currently I handle a CANCEL like an INVITE, which means the CANCEL is sent to user B and not to the vm. How to handle this case? bye, Ingo -- Weitersagen: GMX DSL-Flatrates mit Tempo-Garantie! Ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl

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[Serusers] architecture question on Sending Invite Upon 302 Moved Temp (Redirect)
by mark wehberg 13 Jun '05

13 Jun '05

Hello, There has been other posts as to how to make SER send an INVITE when it gets a 302 Moved Temporaily back from downstream. And it seems that it may be possible to do with AVPOPS, or other scripting methods. The question I have is, what is the reason not to have that functionality built into SER or the option to not "relay back upstream" the 302, but have SER process the message? Is there something I am missing? Call Flow Not Supported Out of the Box INVITE --------------------------> SER SIP SERVER <------------ 100Trying --------------------> INVITE <------------------ 302 Moved <------------------ 302 Moved Instead of INVITE --------------------------> SER SIP SERVER <------------ 100Trying --------------------> INVITE <------------------ 302 Moved --------------------------------------------INVITE ---------------------> Another SIP Server or UA Thanks, Mark

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[Serusers] Problem with cancel a invite
by Norman Brandinger 13 Jun '05

13 Jun '05

Hi, You didn't mention the phones that you are using. We had a similar problem (or at lease similar symptoms) when a call from a SPA-841 is "rejected" by a Polycom-500. In this situation, the SPA-841 still kept ringing. Below is the final in a series of emails I received from Sipura: <snip> You have uncovered a bug in our firmware. In fact all 6xx responses will be ignored by the phone. A temporary fix is available at: http://www.sipura.com/download/temp/phone/spa841-03-01-03-a-0427s-603.zip Thanks for reporting back the problem. - Sipura Support </snip> Regards, Norman Brandinger norm at goes dot com

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[Serusers] Registered Users
by naresh 13 Jun '05

13 Jun '05

Hi, How will I check the users registered with SER Proxy? regards, naresh

1 0

[Serusers] Newbie Questions about SER/serweb
by Anders Hansson 13 Jun '05

13 Jun '05

1 0

[Serusers] rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + ser-0.9.2-fc1.0 CDRTool and it is running fine. (ito.)
by fedora＠voice.dyndns.tv 13 Jun '05

13 Jun '05

rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + ser-0.9.2-fc1.0 CDRTool and it is running fine. (ito.) ser-0.9.2-fc1.0.i386.rpm ser-0.9.2-fc1.0.src.rpm ser-jabber-0.9.2-fc1.0.i386.rpm ser-mysql-0.9.2-fc1.0.i386.rpm ser-radius-0.9.2-fc1.0.i386.rpm ///////////////////////////////////////////// http://www.voice.dyndns.tv sip:admin@iptel.org Saitama Japan Sip EXpress router Support Team voice(a)wmail.plala.or.jp /////////////////////////////////////////////

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Fw: [Serusers] Multiple Domain Problem ser0.9.2
by Rosario Pingaro 12 Jun '05

12 Jun '05

Another little steo ahead; disabling these two parameter: #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") I got the users authenticating only in their domain, so user(a) is only able to autenticate with domain(a). Two problems still persist: 1. two users in different domains but same username get authenticated only the last inserted; 2. the credentials problems persist. Rosario ----- Original Message ----- From: "Rosario Pingaro" <rpingar(a)italycom.it> To: <serusers(a)lists.iptel.org> Sent: Sunday, June 12, 2005 2:54 PM Subject: Re: [Serusers] Multiple Domain Problem ser0.9.2 >I added just to try the domains to alias but it was not the problem, infact >alias only tell ser what domain hav to be included in "myself", I >accomplished that having the domain in my linux hosts file. > > For sure I added the user in the form you specified. So I have all the ser > db lokking good with the domain column filled. > > But the simptoms push me to think that the registrar module is not working > good. In fact also including modparam("registrar", "case_sensitive", 1) > the digest authentication seems case insesitive!!! > > Any other HELP? > > Thanks > > Rosario > > ----- Original Message ----- > From: "Juan Priotti" <jpriotti(a)gmail.com> > To: "Rosario Pingaro" <rpingar(a)italycom.it> > Sent: Sunday, June 12, 2005 12:49 PM > Subject: Re: [Serusers] Multiple Domain Problem ser0.9.2 > > >>I think you would need to add aliases in ser.cfg in this way: >> >> fifo="/tmp/ser_fifo" >> alias=domain1.com domain2.com >> >> so authentication will not fail. >> also when adding users: >> >> serctl add user1(a)domain1.com "passwd" "email" >> >> ----- Original Message ----- >> From: Rosario Pingaro >> To: serusers(a)lists.iptel.org >> Sent: Sunday, June 12, 2005 1:34 PM >> Subject: Fw: [Serusers] Multiple Domain Problem ser0.9.2 >> >> >> I'd like to add that I have added: >> 1. domains using serctl domain add >> 2. added users for domain a and users for domain b >> 3. each user is authenticated without relation with the domain >> 4. each user that is present in domain a and domain b get authentication >> failed (registering in domain a and registrering in doamin b); >> >> thanks again. >> >> Rosario >> >> ----- Original Message ----- >> From: Rosario Pingaro >> To: serusers(a)lists.iptel.org >> Sent: Sunday, June 12, 2005 5:04 PM >> Subject: [Serusers] Multiple Domain Problem ser0.9.2 >> >> >> I'd like to configure SER to host two different domains; the domains are >> into the hst file of the linux machine. >> >> The problem is that SER autheticates the users indifferently from the >> domain >> (I have enabled use_domain on both registrar and userloc), also if an >> user >> is present on both domains i got an authentication failed. >> I don't have alias because the doamin ar einto the linux' host file. >> >> If I remove a domain from the file and don't make an alias I get an error >> about "too many hops" because ser doesn't recognize as myself. >> >> Please may you halp me to debug? >> >> Thanks >> >> Rosario >> >> >> >> These are some part of my ser.cfg >> >> >> >> #debug=6 >> fork=yes >> log_stderror=no >> >> >> check_via=no # (cmd. line: -v) >> dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) >> port=5060 >> children=4 >> fifo="/tmp/ser_fifo" >> fifo_mode=0666 >> fifo_db_url="mysql://ser:hesloimgs@ns2.italycomnet.it/ser" >> listen=213.178.216.45 >> >> modparam("domain", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("domain", "db_mode", 1) >> modparam("domain", "domain_table", "domain") >> modparam("domain", "domain_col", "domain") >> modparam("permissions", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("permissions", "db_mode", 1) >> modparam("permissions", "trusted_table", "trusted") >> modparam("uri_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("usrloc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("usrloc", "db_mode", 1) >> modparam("usrloc", "use_domain",1) >> modparam("auth_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("auth_db", "calculate_ha1", yes) >> modparam("auth_db", "password_column", "password") >> modparam("registrar", "nat_flag", 6) >> modparam("registrar", "use_domain",1) >> modparam("nathelper", "natping_interval", 0) >> modparam("nathelper", "rtpproxy_disable", 1) >> modparam("mediaproxy","natping_interval",30) >> modparam("acc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> modparam("acc", "db_flag", 1) >> modparam("acc", "db_missed_flag", 1) >> modparam("acc", "failed_transactions", 1) >> modparam("acc", "report_cancels", 1) >> modparam("acc", "log_fmt", "cdfimorstup") >> modparam("acc", "log_level", 1) >> modparam("acc", "report_ack", 1) >> modparam("rr", "enable_full_lr", 1) >> modparam("tm", "fr_timer", 40) >> modparam("tm", "fr_inv_timer", 60) >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") >> modparam("tm", "wt_timer", 5) >> modparam("group", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") >> # >> >> route[2] { >> sl_send_reply("100", "ci provo..."); >> if (!search("^Contact:\ +\*") && client_nat_test("7")) { >> setflag(6); >> fix_nated_register(); >> force_rport(); >> }; >> >> if (!www_authorize("", "subscriber")) { >> www_challenge("", "0"); >> break; >> }; >> >> >> route[3] { >> if (client_nat_test("3")){ >> setflag(7); >> force_rport(); >> fix_nated_contact(); >> }; >> if (method=="INVITE" && !allow_trusted()) { >> if (!proxy_authorize("", "subscriber")) { >> proxy_challenge("", "0"); >> break; >> } else if (!check_from()) { >> sl_send_reply("403","Setta From=ID"); >> break; >> }; >> consume_credentials(); >> }; >> lookup("aliases"); >> if (uri!=myself) { >> route(5); >> route(1); >> break; >> }; >> >> if (!lookup("location") && !isflagset(4)) { >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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Re: [Serusers] Multiple Domain Problem ser0.9.2
by Rosario Pingaro 12 Jun '05

12 Jun '05

Another strange thing: I have same user on two different domains: user@domain(a) passw(a) user@domani(b) passw(b) and only user on domain(a) has the credentials. Some time i get authenticated user@domain(b) with pass(b) and I am able to call evean if user@domain(b) doesn't have the appropriate credentials. Another simptoms that SER is not differentiating the domains. Rosario ----- Original Message ----- From: "Juan Priotti" <jpriotti(a)gmail.com> To: "Rosario Pingaro" <rpingar(a)italycom.it> Sent: Sunday, June 12, 2005 12:49 PM Subject: Re: [Serusers] Multiple Domain Problem ser0.9.2 >I think you would need to add aliases in ser.cfg in this way: > > fifo="/tmp/ser_fifo" > alias=domain1.com domain2.com > > so authentication will not fail. > also when adding users: > > serctl add user1(a)domain1.com "passwd" "email" > > ----- Original Message ----- > From: Rosario Pingaro > To: serusers(a)lists.iptel.org > Sent: Sunday, June 12, 2005 1:34 PM > Subject: Fw: [Serusers] Multiple Domain Problem ser0.9.2 > > > I'd like to add that I have added: > 1. domains using serctl domain add > 2. added users for domain a and users for domain b > 3. each user is authenticated without relation with the domain > 4. each user that is present in domain a and domain b get authentication > failed (registering in domain a and registrering in doamin b); > > thanks again. > > Rosario > > ----- Original Message ----- > From: Rosario Pingaro > To: serusers(a)lists.iptel.org > Sent: Sunday, June 12, 2005 5:04 PM > Subject: [Serusers] Multiple Domain Problem ser0.9.2 > > > I'd like to configure SER to host two different domains; the domains are > into the hst file of the linux machine. > > The problem is that SER autheticates the users indifferently from the > domain > (I have enabled use_domain on both registrar and userloc), also if an user > is present on both domains i got an authentication failed. > I don't have alias because the doamin ar einto the linux' host file. > > If I remove a domain from the file and don't make an alias I get an error > about "too many hops" because ser doesn't recognize as myself. > > Please may you halp me to debug? > > Thanks > > Rosario > > > > These are some part of my ser.cfg > > > > #debug=6 > fork=yes > log_stderror=no > > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=4 > fifo="/tmp/ser_fifo" > fifo_mode=0666 > fifo_db_url="mysql://ser:hesloimgs@ns2.italycomnet.it/ser" > listen=213.178.216.45 > > modparam("domain", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > modparam("permissions", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > modparam("uri_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("usrloc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "use_domain",1) > modparam("auth_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("registrar", "nat_flag", 6) > modparam("registrar", "use_domain",1) > modparam("nathelper", "natping_interval", 0) > modparam("nathelper", "rtpproxy_disable", 1) > modparam("mediaproxy","natping_interval",30) > modparam("acc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "report_ack", 1) > modparam("rr", "enable_full_lr", 1) > modparam("tm", "fr_timer", 40) > modparam("tm", "fr_inv_timer", 60) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > modparam("tm", "wt_timer", 5) > modparam("group", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > # > > route[2] { > sl_send_reply("100", "ci provo..."); > if (!search("^Contact:\ +\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > > > route[3] { > if (client_nat_test("3")){ > setflag(7); > force_rport(); > fix_nated_contact(); > }; > if (method=="INVITE" && !allow_trusted()) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > } else if (!check_from()) { > sl_send_reply("403","Setta From=ID"); > break; > }; > consume_credentials(); > }; > lookup("aliases"); > if (uri!=myself) { > route(5); > route(1); > break; > }; > > if (!lookup("location") && !isflagset(4)) { > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

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Re: [Serusers] Multiple Domain Problem ser0.9.2
by Rosario Pingaro 12 Jun '05

12 Jun '05

I added just to try the domains to alias but it was not the problem, infact alias only tell ser what domain hav to be included in "myself", I accomplished that having the domain in my linux hosts file. For sure I added the user in the form you specified. So I have all the ser db lokking good with the domain column filled. But the simptoms push me to think that the registrar module is not working good. In fact also including modparam("registrar", "case_sensitive", 1) the digest authentication seems case insesitive!!! Any other HELP? Thanks Rosario ----- Original Message ----- From: "Juan Priotti" <jpriotti(a)gmail.com> To: "Rosario Pingaro" <rpingar(a)italycom.it> Sent: Sunday, June 12, 2005 12:49 PM Subject: Re: [Serusers] Multiple Domain Problem ser0.9.2 >I think you would need to add aliases in ser.cfg in this way: > > fifo="/tmp/ser_fifo" > alias=domain1.com domain2.com > > so authentication will not fail. > also when adding users: > > serctl add user1(a)domain1.com "passwd" "email" > > ----- Original Message ----- > From: Rosario Pingaro > To: serusers(a)lists.iptel.org > Sent: Sunday, June 12, 2005 1:34 PM > Subject: Fw: [Serusers] Multiple Domain Problem ser0.9.2 > > > I'd like to add that I have added: > 1. domains using serctl domain add > 2. added users for domain a and users for domain b > 3. each user is authenticated without relation with the domain > 4. each user that is present in domain a and domain b get authentication > failed (registering in domain a and registrering in doamin b); > > thanks again. > > Rosario > > ----- Original Message ----- > From: Rosario Pingaro > To: serusers(a)lists.iptel.org > Sent: Sunday, June 12, 2005 5:04 PM > Subject: [Serusers] Multiple Domain Problem ser0.9.2 > > > I'd like to configure SER to host two different domains; the domains are > into the hst file of the linux machine. > > The problem is that SER autheticates the users indifferently from the > domain > (I have enabled use_domain on both registrar and userloc), also if an user > is present on both domains i got an authentication failed. > I don't have alias because the doamin ar einto the linux' host file. > > If I remove a domain from the file and don't make an alias I get an error > about "too many hops" because ser doesn't recognize as myself. > > Please may you halp me to debug? > > Thanks > > Rosario > > > > These are some part of my ser.cfg > > > > #debug=6 > fork=yes > log_stderror=no > > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > port=5060 > children=4 > fifo="/tmp/ser_fifo" > fifo_mode=0666 > fifo_db_url="mysql://ser:hesloimgs@ns2.italycomnet.it/ser" > listen=213.178.216.45 > > modparam("domain", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > modparam("permissions", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("permissions", "db_mode", 1) > modparam("permissions", "trusted_table", "trusted") > modparam("uri_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("usrloc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("usrloc", "db_mode", 1) > modparam("usrloc", "use_domain",1) > modparam("auth_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("registrar", "nat_flag", 6) > modparam("registrar", "use_domain",1) > modparam("nathelper", "natping_interval", 0) > modparam("nathelper", "rtpproxy_disable", 1) > modparam("mediaproxy","natping_interval",30) > modparam("acc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > modparam("acc", "db_flag", 1) > modparam("acc", "db_missed_flag", 1) > modparam("acc", "failed_transactions", 1) > modparam("acc", "report_cancels", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "log_level", 1) > modparam("acc", "report_ack", 1) > modparam("rr", "enable_full_lr", 1) > modparam("tm", "fr_timer", 40) > modparam("tm", "fr_inv_timer", 60) > modparam("tm", "fr_inv_timer_avp", "inv_timeout") > modparam("tm", "wt_timer", 5) > modparam("group", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") > # > > route[2] { > sl_send_reply("100", "ci provo..."); > if (!search("^Contact:\ +\*") && client_nat_test("7")) { > setflag(6); > fix_nated_register(); > force_rport(); > }; > > if (!www_authorize("", "subscriber")) { > www_challenge("", "0"); > break; > }; > > > route[3] { > if (client_nat_test("3")){ > setflag(7); > force_rport(); > fix_nated_contact(); > }; > if (method=="INVITE" && !allow_trusted()) { > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "0"); > break; > } else if (!check_from()) { > sl_send_reply("403","Setta From=ID"); > break; > }; > consume_credentials(); > }; > lookup("aliases"); > if (uri!=myself) { > route(5); > route(1); > break; > }; > > if (!lookup("location") && !isflagset(4)) { > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

Fw: [Serusers] Multiple Domain Problem ser0.9.2
by Rosario Pingaro 12 Jun '05

12 Jun '05

I'd like to add that I have added: 1. domains using serctl domain add 2. added users for domain a and users for domain b 3. each user is authenticated without relation with the domain 4. each user that is present in domain a and domain b get authentication failed (registering in domain a and registrering in doamin b); thanks again. Rosario ----- Original Message ----- From: Rosario Pingaro To: serusers(a)lists.iptel.org Sent: Sunday, June 12, 2005 5:04 PM Subject: [Serusers] Multiple Domain Problem ser0.9.2 I'd like to configure SER to host two different domains; the domains are into the hst file of the linux machine. The problem is that SER autheticates the users indifferently from the domain (I have enabled use_domain on both registrar and userloc), also if an user is present on both domains i got an authentication failed. I don't have alias because the doamin ar einto the linux' host file. If I remove a domain from the file and don't make an alias I get an error about "too many hops" because ser doesn't recognize as myself. Please may you halp me to debug? Thanks Rosario These are some part of my ser.cfg #debug=6 fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:hesloimgs@ns2.italycomnet.it/ser" listen=213.178.216.45 modparam("domain", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("permissions", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("uri_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("usrloc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "use_domain",1) modparam("auth_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain",1) modparam("nathelper", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) modparam("mediaproxy","natping_interval",30) modparam("acc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "report_ack", 1) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 60) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "wt_timer", 5) modparam("group", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") # route[2] { sl_send_reply("100", "ci provo..."); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; route[3] { if (client_nat_test("3")){ setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if (!check_from()) { sl_send_reply("403","Setta From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location") && !isflagset(4)) { -------------------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Multiple Domain Problem ser0.9.2
by Rosario Pingaro 12 Jun '05

12 Jun '05

I'd like to configure SER to host two different domains; the domains are into the hst file of the linux machine. The problem is that SER autheticates the users indifferently from the domain (I have enabled use_domain on both registrar and userloc), also if an user is present on both domains i got an authentication failed. I don't have alias because the doamin ar einto the linux' host file. If I remove a domain from the file and don't make an alias I get an error about "too many hops" because ser doesn't recognize as myself. Please may you halp me to debug? Thanks Rosario These are some part of my ser.cfg #debug=6 fork=yes log_stderror=no check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:hesloimgs@ns2.italycomnet.it/ser" listen=213.178.216.45 modparam("domain", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") modparam("permissions", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("uri_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("usrloc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "use_domain",1) modparam("auth_db", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain",1) modparam("nathelper", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) modparam("mediaproxy","natping_interval",30) modparam("acc", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 1) modparam("acc", "failed_transactions", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "log_level", 1) modparam("acc", "report_ack", 1) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 40) modparam("tm", "fr_inv_timer", 60) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("tm", "wt_timer", 5) modparam("group", "db_url", "mysql://ser:heslo@xxx.xxxxxx.it/ser") # route[2] { sl_send_reply("100", "ci provo..."); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; route[3] { if (client_nat_test("3")){ setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if (!check_from()) { sl_send_reply("403","Setta From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location") && !isflagset(4)) {

1 0

[Serusers] different gateways for different domain
by Rosario Pingaro 12 Jun '05

12 Jun '05

I'd like to differentiate the gateways based on the uri domain, I tought that if (uri==domain1.com) ....... could help but it doesn't. How to accomplish this? thanks in advance. Rosario

1 1

[Serusers] set_sattr?
by Chris St Denis 12 Jun '05

12 Jun '05

Like this mailing list entry from Dec 2004, I need to have the INVITE timer wait longer when calling some numbers than others. http://lists.iptel.org/pipermail/serusers/2004-December/013701.html The response says to use set_sattr("callee_fr_inv_timer", "123"); I've upgraded to ser .9.0 (unstable?) but it still doesn't seem to exist in that version. What version do I need for this function? Will it be available in .9.2 stable when it is released?

3 2

[Serusers] DNS Resolution of SDP-"Connection Address" at SER
by Gerhard Zweimüller 12 Jun '05

12 Jun '05

Hi Serusers, I have a problem with DNS resolution in SIP/SDP "Connection Information" / "ConnectionAddress". Maybe somebody can help me: In the network we use SIP-UAs that are integrated into ADSL-Modems from Allied Telesyn (AT RG 634). The SIP-client itself works OK. But when the unit is given a system-name in the configuration, it will always use its DNS-name instead of its IP-addr as "Connection Address" in the "Connection Information" in SDP in INVITE and corresponding OK messages. Now we want to add SIP-UAs that are NOT capable of resolving DNS-Names in the "Connection Address" field. Alle messages pass through SER of course and the SDP-part so far remains unchanged. Now my question to the list: Is there a simple way of forcing SER to do the DNS-resolution of in incoming message, put in the IP-address in "Connection Address" and forward it to the other UA? Thanks a lot in advance! Gerhard

3 2

[Serusers] SER + MySql (beginner)
by Valério Vendrame Costa 12 Jun '05

12 Jun '05

Hi, my name is Valerio and I'm from Brazil... I'm an beginning user of SER... Folders Structure: /usr/local/sbin/ -> ser and serctl. /usr/local/etc/ser/ser.cfg /usr/local/lib/ser/modules/ -> mysql.so, usrloc.so, *.so... /usr/local/mysql/bin/ -> mysqld_safe. /usr/lib/ -> libmysqlclient.so.10. Well... I'm using SER 0.8.14_linux_i386 and mysql-standard-4.1.12-pc-linux-gnu-i686. When I use "original" ser.cfg, SER runs ok... When I use mysql.so module i have problems and SER don't start... I initiate mysql: /usr/local/mysql/bin/mysqld_safe & <enter> I initiate ser: /usr/local/sbin/serctl start <enter> (if load with original ser.cfg its ok! But when I enable access to MySql in ser.cfg don't run...) Somebody can help me? Thanks, Valerio. _________________________________________________________________ MSN Messenger: converse online com seus amigos . http://messenger.msn.com.br

2 1

[Serusers] mediaproxy / rtp session
by harry gaillac 11 Jun '05

11 Jun '05

Hello serusers, Don't reply to mails i 've sent. I solved the problem by rewriting onreply_route[x] I add search for Contact-type instead of Contact-length because of SDP message is return with "200 OK". Harry I get this now : ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Fwd: Re: [Serusers] VoIp Billing Solution SIP server compatible!!!]
by Iqbal 11 Jun '05

11 Jun '05

Hi This was my pseudo code, which I used CDRTool as a base, and then together with what serweb had I created a complete tool, since then I have changed it alot to do diff prices per customer type etc. As for tax scripts, thats just 17.5% here in the UK, if it varies youy can just add a field per rate defined Iqbal

2 1

[Serusers] How to remove the 8 requirement in serweb?
by Ronald Rebideau 11 Jun '05

11 Jun '05

I swear I've googled it and searched the archives but I can't seem to find the answer. I realize this is going to be a 'Doh' moment but can some throw me a bone here? I just want to be able to add regular phone numbers via serweb.

1 0

[Serusers] mediaproxy / rtp session
by harry gaillac 11 Jun '05

11 Jun '05

Hello, According to ethereal sip_call file I have a problem with ser.cfg When callee reply status 200 OK to caller (line 14,15 of sip_call) with SIP/SDP message address:port are not good ? route(1) in ser.cfg push message with code 200 in mediaproxy!? How can i fix this problem ? Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Cisco ATA and SER
by M. Ehsanul Karim 11 Jun '05

11 Jun '05

I am using Cisco ATA to register to a SER server . SER forwards to an Asterisk box. But the Cisco does not get any dialtone even if it registers. It works fine with Sipura. Can anyone share their CIsco and SER config with me. Thanks for your time. Ehsan

1 0

[Serusers] mediaproxy/rtp session
by harry gaillac 11 Jun '05

11 Jun '05

Hello serusers, I installed the lastest ser release (unstable) + mediaproxy on the same box and clients are behind nat. mediaproxy handle rtp session until idleTimeout = 60 according to mediaproxy.ini file. Status session is never "active". Is it bad command requests ? Regards Harry Jun 11 14:43:46 serveur1 mediaproxy[6141]: command request 4eee61c5-3a7bdd4f-1972bc52(a)192.168.0.21 192.168.0.21:35000:audio 192.168.0.21 nxs.yi.org local nxs.yi.org local PolycomSoundPointIP-SPIP_300-UA/1.5.2.0054 info=from:bob@nxs.yi.org,to:84@nxs.yi.org,fromtag:921C3699-357539BC,totag: Jun 11 14:43:46 serveur1 mediaproxy[6141]: session 4eee61c5-3a7bdd4f-1972bc52(a)192.168.0.21: started. listening on 80.119.10.135:35006 Jun 11 14:43:46 serveur1 mediaproxy[6141]: command execution time: 1.22 ms Jun 11 14:43:46 serveur1 mediaproxy[6141]: command request 4eee61c5-3a7bdd4f-1972bc52(a)192.168.0.21 192.168.0.21:35000:audio 192.168.0.21 nxs.yi.org local nxs.yi.org local PolycomSoundPointIP-SPIP_300-UA/1.5.2.0054 info=from:bob@nxs.yi.org,to:84@nxs.yi.org,fromtag:921C3699-357539BC,totag: Jun 11 14:43:46 serveur1 mediaproxy[6141]: command execution time: 0.31 ms Jun 11 14:43:53 serveur1 mediaproxy[6141]: command status Jun 11 14:43:53 serveur1 mediaproxy[6141]: command execution time: 0.33 ms Jun 11 14:44:45 serveur1 mediaproxy[6141]: session 4eee61c5-3a7bdd4f-1972bc52(a)192.168.0.21: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) Jun 11 14:44:45 serveur1 mediaproxy[6141]: session 4eee61c5-3a7bdd4f-1972bc52(a)192.168.0.21: ended (did timeout). ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] Out-of-office-reply: Serusers Digest, Vol 26, Issue 17
by David.Monahan＠aepona.com 11 Jun '05

11 Jun '05

I will be out of the office until Monday 27th June. For urgent queries please contact Judith Russell (judith.russell(a)aepona.com).

1 0

[Serusers] error when starting SER
by Kamran Ahmad 11 Jun '05

11 Jun '05

hello here is /var/log/messages. i am unable to stop SER. even i tried to killall ser but not working. ser process is always in memory and no successfull registration. serctl moni is not working what is the wrong with my SER-0.9.0 Jun 4 07:04:02 voice ser: WARNING: fix_socket_list: could not rev. resolve IP Jun 4 07:04:02 voice ser: WARNING: fix_socket_list: could not rev. resolve IP Jun 4 07:04:02 voice ser[25577]: Maxfwd module- initializing Jun 4 07:04:02 voice ser[25577]: INFO: udp_init: SO_RCVBUF is initially 110592 Jun 4 07:04:02 voice ser[25577]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 4 07:04:02 voice ser[25577]: INFO: udp_init: SO_RCVBUF is initially 110592 Jun 4 07:04:02 voice ser[25577]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 4 07:04:02 voice ser[25577]: ERROR: tcp_init: bind(7, 0x8146114, 16) on 127.0.0.1: Address already in use __________________________________ Discover Yahoo! Get on-the-go sports scores, stock quotes, news and more. Check it out! http://discover.yahoo.com/mobile.html

1 0

[Serusers] Radius question
by Alberto Cruz 11 Jun '05

11 Jun '05

Hi guys, I have a doubt. Is it possible to use radius related modules with out using db related modules at all? Which are going to be the side effects If I don't use db modules? For example I tried to load mediaproxy module and it is claiming for commands like "is_uri_host_local()" or "is_from_local()" in order to be load it but these commands are only available if I load the db modules. Who can help me to clarify the relationship between radius modules, mediaproxy module and db modules? Regards Alberto Cruz

1 0

[Serusers] mediaproxy server doesn't show rtp connections
by Rosario Pingaro 10 Jun '05

10 Jun '05

I have configured two server one with ser and dispacher and the second with mediaproxy server. I didn't set the dns records but i choosed to use a default proxy. Everything seems to work fine: 1. into the dispacher machine i can log the closing of the media session 2. the audio from and to private ip is fine BUT when I execute sessions.py on the server machine i get NO RTP SESSIONS, even if i am still calling! the same if i use the script to generat the web page. what is going on? Rosario

1 0

[Serusers] Linksys PAP2NA
by gsalvucci＠ing.unrc.edu.ar 10 Jun '05

10 Jun '05

Hi, I have a problem with the device Linksys PAP2NA. I'm using the mediaproxy server and the mediaproxy module with ser.cfg from mediaproxy modules. I have Linksys behind a NAT and I comunicate with a x-lite user agent using a public IP. I can get outgoing comunications but I can't get incoming comunications because the server add so many via headers with the server IP. As a result, I get the response "too many hops". I have proved replacing the linksys with a x-lite user agent and it works. Anyone knows what is the problem? Thanks in advanced. Gustavo Salvucci Rio Cuarto. Cordoba. Argentina. ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

1 0

[Users] Alien Probe :)
by Bogdan-Andrei Iancu 10 Jun '05

10 Jun '05

probing list

1 0

[Serusers] How to execute failure_route block by the same child process.
by Dmitry Isakbayev 10 Jun '05

10 Jun '05

Hello, I am trying to implement sequential forking using to_on_failure and failure_route blocks. Is there a way to configure SER or tm module to execute the 'failure-route' blocks using the same process as was used for executing the main route script? If it was possible, I would not need to share state information between different SER processes Thank you in advance, Dmitry

4 4

[Serusers] Load Ballancing
by TCCHAN 10 Jun '05

10 Jun '05

Dear All, Does anyone know how to load balance SER call control signaling. I meant how to get multiple copies of SER running on different machine. Is it through Virtual Linux Server or something else. Please advise. Thank for any help available. Regards, TC Chan

4 3

[Serusers] Single Registration Restriction
by Darren Nay 10 Jun '05

10 Jun '05

Hey all, Can anyone tell me a good and reliable way to restrict registrations to 1 per username? I was thinking that I could do something like this: if (method=="REGISTER") { if (!www_authorize("proxy.mydomain.net", "subscriber")) { www_challenge("proxy.mydomain.net", "0"); break; }; }; if (!lookup("location")) { save("location"); } else { ## instead of saving with a 200 OK - send error ## Not sure the best error to return here sl_send_reply("403","Forbidden"); } break; }; The problem is that we use static routes for voicemail, and so there will always be at least one static route in the locations table for each of our customers and thus the method above won't work. Is there a way that I can not only check whether a location exists, but also do a Search() against the URI's of the returned destination set? Or is there another (possibly better) way to do this? Thanks for your help!! Darren Nay Ionosphere, Inc. VoIP Network Development dnay(a)ionosphere.net

2 1

[Serusers] error while compiling radius modules
by Kamran Ahmad 10 Jun '05

10 Jun '05

hi all can u tell me what is the wrong in my mechine i am using radiusclient-ng make[1]: Entering directory `/ser-0.9.0/modules/auth_radius' ../../Makefile.rules:77: authorize.d: No such file or directory ../../Makefile.rules:77: authrad_mod.d: No such file or directory ../../Makefile.rules:77: sterman.d: No such file or directory make[1]: Leaving directory `/ser-0.9.0/modules/auth_radius' make[1]: Entering directory `/ser-0.9.0/modules/auth_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c authorize.c -o authorize.o gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c authrad_mod.c -o authrad_mod.o authrad_mod.c: In function `mod_init': authrad_mod.c:110: error: `DICT_VENDOR' undeclared (first use in this function) authrad_mod.c:110: error: (Each undeclared identifier is reported only once authrad_mod.c:110: error: for each function it appears in.) authrad_mod.c:110: error: `vend' undeclared (first use in this function) authrad_mod.c:134: warning: assignment makes pointer from integer without a cast authrad_mod.c:139: error: too many arguments to function `rc_conf_str' authrad_mod.c:139: error: too many arguments to function `rc_read_dictionary' authrad_mod.c:144: warning: implicit declaration of function `rc_dict_findvend' authrad_mod.c:159: error: too many arguments to function `rc_dict_findattr' authrad_mod.c:159: error: too many arguments to function `rc_dict_findval' make[1]: *** [authrad_mod.o] Error 1 make[1]: Leaving directory `/ser-0.9.0/modules/auth_radius' make[1]: Entering directory `/ser-0.9.0/modules/group_radius' ../../Makefile.rules:77: group.d: No such file or directory ../../Makefile.rules:77: grouprad_mod.d: No such file or directory make[1]: Leaving directory `/ser-0.9.0/modules/group_radius' make[1]: Entering directory `/ser-0.9.0/modules/group_radius' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mtune=athlon -I/opt/axxany/siprouter/radiusclient/include -DNAME='"ser"' -DVERSION='"0.9.1"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.4"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -c group.c -o group.o group.c: In function `radius_is_user_in': group.c:194: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:194: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:194: error: too many arguments to function `rc_avpair_add' group.c:203: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:203: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:203: error: too many arguments to function `rc_avpair_add' group.c:209: warning: passing arg 2 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: warning: passing arg 3 of `rc_avpair_add' makes pointer from integer without a cast group.c:209: warning: passing arg 4 of `rc_avpair_add' makes integer from pointer without a cast group.c:209: error: too many arguments to function `rc_avpair_add' group.c:215: warning: passing arg 1 of `rc_auth' makes integer from pointer without a cast group.c:215: warning: passing arg 3 of `rc_auth' from incompatible pointer type group.c:215: warning: passing arg 4 of `rc_auth' from incompatible pointer type group.c:215: error: too many arguments to function `rc_auth' make[1]: *** [group.o] Error 1 make[1]: Leaving directory `/ser-0.9.0/modules/group_radius' __________________________________________________ Do You Yahoo!? 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1 0

[Serusers] usrloc bug report
by Java Rockx 10 Jun '05

10 Jun '05

Hi All. Here's a bit more info on the usrloc "flags" bug. This register message was flagged at NATed eventhough my "nat_flag" was not set. U 2005/06/10 11:01:27.853890 71.9.17.51:5060 <http://71.9.17.51:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> REGISTER sip:sipdev.mycompany.net <http://sipdev.mycompany.net> SIP/2.0. Via: SIP/2.0/UDP 71.9.17.51:5060 <http://71.9.17.51:5060> ;branch=z9hG4bK1106535059. From: Paul <sip:3215590000@sipdev.company.net>;tag=3843647257. To: Paul <sip:3215590000@sipdev.company.net>. Call-ID: 1846387046(a)71.9.17.51. CSeq: 778 REGISTER. Contact: <sip:3215590000@71.9.17.51:5060>. Authorization: Digest username="3215590000", realm="sipdev.mycompany.net<http://sipdev.mycompany.net>", nonce="42a9ac7341e18f72830c3cd553fecfbd0c9fc6", uri="sip: sipdev.company.net:5060 <http://sipdev.company.net:5060>", response="5443dedef7bfa7fb4a033af66a2392c", algorithm=MD5, cnonce="3dc4445e42acd47873ca563a9942104a", qop=auth, nc=00000193. max-forwards: 70. expires: 900. Content-Length: 0. Another thing I see is that all usrloc records end up having the flag column set to 1 wheather the client is NATed or not. I'm setting the registrar modules nat_flag using client_nat_test("7") when processing REGISTER messages, however usrloc is not correctly setting the NAT_FLAG. Regards, Paul

2 2

[Serusers] usrloc bug report
by Java Rockx 10 Jun '05

10 Jun '05

Hi All. I've identified a bug in the usrloc module (I think). This can be reproduced. When a non-NATed client registers, usrloc will write an entry to the MySQL location table. If I manually change the "flags" column from 0 to 1 (which indicates a NATed client) and then restart SER, the next time the non-NATed client registers and usrloc refreshes the location table it does not update the flags column. Regards, Paul

2 3

[Serusers] usrloc bug report
by Java Rockx 10 Jun '05

10 Jun '05

Hi All. I forgot to mention that this bug exists in ser-0.9.2 checked out of CVS last week. http://lists.iptel.org/pipermail/serusers/2005-June/020560.html Regards, Paul

1 0

[Serusers] serweb pconnect error
by Pacific 10 Jun '05

10 Jun '05

Hi, I have a problem with serweb. It gives this error if I try to open the page http://hostofmyserver/serweb/admin/index.php Database error: pconnect(hostofmyserver, ser, $Password) failed. MySQL Error: () Session halted. but I'm sure ser and mysql are ok because I tested them with kphone and they worked fine. I followed the steps at this page because I have fedora core 2 http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/… Somebody can help me? tnks ___________________________________ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it

2 1

RE: [Serusers] Problems with version SER 0.9
by Simith Nambiar 10 Jun '05

10 Jun '05

Hello Jesus, Check whether the IP Address in your Ser.cfg file is correct, If that wasn't the problem , you can download the config files from www.onsip.org <http://www.onsip.org> , There are some Basic configuration files out there, for beginners like us :-) it worked for me , Good luck, Simith -----Original Message----- From: Jesús M. Movilla [mailto:jmms03@tid.es] Sent: Friday, June 10, 2005 5:38 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Problems with version SER 0.9 Hi to all, I took the sources of SER version 0.9. and I compiled them. I installed this SER version but when I tried to start it up I got this error: textops - initializing 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) qm_free(0x80fbe60, 0x8130d58), called from maxfwd.c: fixup_maxfwd_header(122) 0(0) qm_free: freeing frag. 0x8130d40 alloc'ed from cfg.lex: addstr(572) 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) qm_free(0x80fbe60, 0x8130f90), called from sl.c: fixup_sl_send_reply(142) 0(0) qm_free: freeing frag. 0x8130f78 alloc'ed from cfg.lex: addstr(572) 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) qm_free(0x80fbe60, 0x81312d4), called from sl.c: fixup_sl_send_reply(142) 0(0) qm_free: freeing frag. 0x81312bc alloc'ed from cfg.lex: addstr(572) ERROR: error -478 while trying to fix configuration 0(0) qm_free(0x80fbe60, 0x813096c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130954 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x813087c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130864 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x403b6400), called from ul_callback.c: destroy_ulcb_list(75) 0(0) qm_free: freeing frag. 0x403b63e8 alloc'ed from ul_callback.c: init_ulcb_list(50) 0(0) qm_free(0x80fbe60, 0x813078c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130774 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x813069c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130684 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x81305ac), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130594 alloc'ed from sr_module.c: register_module(136) 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) qm_free(0x401ff000, 0x40236070), called from h_table.c: free_hash_table(309) 0(0) qm_free: freeing frag. 0x40236058 alloc'ed from h_table.c: init_hash_table(323) 0(0) DEBUG: tm_shutdown : releasing timers 0(0) qm_free(0x401ff000, 0x403b60e0), called from timer.c: free_timer_table(551) 0(0) qm_free: freeing frag. 0x403b60c8 alloc'ed from timer.c: tm_init_timers(515) 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) qm_free(0x401ff000, 0x403b60a0), called from lock.c: lock_cleanup(205) 0(0) qm_free: freeing frag. 0x403b6088 alloc'ed from lock.c: lock_initialize(99) 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) qm_free(0x401ff000, 0x403b63c8), called from t_hooks.c: destroy_tmcb_lists(83) 0(0) qm_free: freeing frag. 0x403b63b0 alloc'ed from t_hooks.c: init_tmcb_lists(58) 0(0) qm_free(0x401ff000, 0x403b6368), called from t_stats.c: free_tm_stats(223) 0(0) qm_free: freeing frag. 0x403b6350 alloc'ed from t_stats.c: init_tm_stats(185) 0(0) qm_free(0x401ff000, 0x403b6308), called from t_stats.c: free_tm_stats(225) 0(0) qm_free: freeing frag. 0x403b62f0 alloc'ed from t_stats.c: init_tm_stats(178) 0(0) qm_free(0x401ff000, 0x403b62a8), called from t_stats.c: free_tm_stats(227) 0(0) qm_free: freeing frag. 0x403b6290 alloc'ed from t_stats.c: init_tm_stats(171) 0(0) qm_free(0x401ff000, 0x403b6250), called from t_stats.c: free_tm_stats(228) 0(0) qm_free: freeing frag. 0x403b6238 alloc'ed from t_stats.c: init_tm_stats(163) 0(0) DEBUG: tm_shutdown : done 0(0) qm_free(0x80fbe60, 0x81304bc), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x81304a4 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x40235bfc), called from sl_stats.c: sl_stats_destroy(164) 0(0) qm_free: freeing frag. 0x40235be4 alloc'ed from sl_stats.c: init_sl_stats(179) 0(0) qm_free(0x401ff000, 0x4023603c), called from sl_funcs.c: sl_shutdown(99) 0(0) qm_free: freeing frag. 0x40236024 alloc'ed from sl_funcs.c: sl_startup(82) 0(0) qm_free(0x80fbe60, 0x81303cc), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x81303b4 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x40233474), called from tcp_main.c: destroy_tcp(1367) 0(0) qm_free: freeing frag. 0x4023345c alloc'ed from tcp_main.c: init_tcp(1303) 0(0) qm_free(0x401ff000, 0x402334dc), called from tcp_main.c: destroy_tcp(1371) 0(0) qm_free: freeing frag. 0x402334c4 alloc'ed from tcp_main.c: init_tcp(1326) 0(0) qm_free(0x401ff000, 0x4023450c), called from tcp_main.c: destroy_tcp(1375) 0(0) qm_free: freeing frag. 0x402344f4 alloc'ed from tcp_main.c: init_tcp(1338) 0(0) qm_free(0x401ff000, 0x402334a8), called from tcp_main.c: destroy_tcp(1380) 0(0) qm_free: freeing frag. 0x40233490 alloc'ed from tcp_main.c: init_tcp(1315) 0(0) qm_free(0x401ff000, 0x40233440), called from timer.c: destroy_timer(80) 0(0) qm_free: freeing frag. 0x40233428 alloc'ed from timer.c: init_timer(57) 0(0) qm_free(0x80fbe60, 0x8134104), called from timer.c: destroy_timer(89) 0(0) qm_free: freeing frag. 0x81340ec alloc'ed from timer.c: register_timer(104) 0(0) qm_free(0x80fbe60, 0x8133fbc), called from timer.c: destroy_timer(89) 0(0) qm_free: freeing frag. 0x8133fa4 alloc'ed from timer.c: register_timer(104) 0(0) qm_free(0x80fbe60, 0x81344cc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81344b4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813448c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134474 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813444c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134434 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813440c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81343f4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x81343cc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81343b4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813438c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134374 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813434c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134334 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8134044), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x813402c alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133f7c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133f64 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133f3c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133f24 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133efc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133ee4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133ebc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133ea4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133d3c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133d24 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813430c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81342f4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81342cc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81342b4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813428c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134274 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813424c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134234 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813420c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81341f4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81341cc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81341b4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813418c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134174 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8134004), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133fec alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133e7c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133e64 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133e3c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133e24 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133dfc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133de4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133dbc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133da4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133cfc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133ce4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81340c4), called from script_cb.c: destroy_script_cb(81) 0(0) qm_free: freeing frag. 0x81340ac alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x80fbe60, 0x8133d7c), called from script_cb.c: destroy_script_cb(81) 0(0) qm_free: freeing frag. 0x8133d64 alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x80fbe60, 0x8134084), called from script_cb.c: destroy_script_cb(83) 0(0) qm_free: freeing frag. 0x813406c alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x401ff000, 0x4023553c), called from main.c: cleanup(373) 0(0) qm_free: freeing frag. 0x40235524 alloc'ed from main.c: main(1534) 0(2180) shm_mem_destroy 0(2180) destroying the shared memory lock The ser.cfg that I have used is the same included in the compilation except for the debugs to get activated. Next I include it in case you see something rare: Any help or idea is very very welcome :-) Thank you very much Jesús ********************************************************************** # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (src_ip==193.175.135.0/24){ force_send_socket(smaug:5080); forward(193.175.135.179); break; } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

1 0

RE: [Serusers] Re: SEMS not working, problem solved, thank you
by Llanos Serna García-Conde 10 Jun '05

10 Jun '05

1 0

[Serusers] Problems with version SER 0.9
by Jesús M. Movilla 10 Jun '05

10 Jun '05

Hi to all, I took the sources of SER version 0.9. and I compiled them. I installed this SER version but when I tried to start it up I got this error: textops - initializing 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) qm_free(0x80fbe60, 0x8130d58), called from maxfwd.c: fixup_maxfwd_header(122) 0(0) qm_free: freeing frag. 0x8130d40 alloc'ed from cfg.lex: addstr(572) 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) qm_free(0x80fbe60, 0x8130f90), called from sl.c: fixup_sl_send_reply(142) 0(0) qm_free: freeing frag. 0x8130f78 alloc'ed from cfg.lex: addstr(572) 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) qm_free(0x80fbe60, 0x81312d4), called from sl.c: fixup_sl_send_reply(142) 0(0) qm_free: freeing frag. 0x81312bc alloc'ed from cfg.lex: addstr(572) ERROR: error -478 while trying to fix configuration 0(0) qm_free(0x80fbe60, 0x813096c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130954 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x813087c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130864 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x403b6400), called from ul_callback.c: destroy_ulcb_list(75) 0(0) qm_free: freeing frag. 0x403b63e8 alloc'ed from ul_callback.c: init_ulcb_list(50) 0(0) qm_free(0x80fbe60, 0x813078c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130774 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x813069c), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130684 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x80fbe60, 0x81305ac), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x8130594 alloc'ed from sr_module.c: register_module(136) 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) qm_free(0x401ff000, 0x40236070), called from h_table.c: free_hash_table(309) 0(0) qm_free: freeing frag. 0x40236058 alloc'ed from h_table.c: init_hash_table(323) 0(0) DEBUG: tm_shutdown : releasing timers 0(0) qm_free(0x401ff000, 0x403b60e0), called from timer.c: free_timer_table(551) 0(0) qm_free: freeing frag. 0x403b60c8 alloc'ed from timer.c: tm_init_timers(515) 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) qm_free(0x401ff000, 0x403b60a0), called from lock.c: lock_cleanup(205) 0(0) qm_free: freeing frag. 0x403b6088 alloc'ed from lock.c: lock_initialize(99) 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) qm_free(0x401ff000, 0x403b63c8), called from t_hooks.c: destroy_tmcb_lists(83) 0(0) qm_free: freeing frag. 0x403b63b0 alloc'ed from t_hooks.c: init_tmcb_lists(58) 0(0) qm_free(0x401ff000, 0x403b6368), called from t_stats.c: free_tm_stats(223) 0(0) qm_free: freeing frag. 0x403b6350 alloc'ed from t_stats.c: init_tm_stats(185) 0(0) qm_free(0x401ff000, 0x403b6308), called from t_stats.c: free_tm_stats(225) 0(0) qm_free: freeing frag. 0x403b62f0 alloc'ed from t_stats.c: init_tm_stats(178) 0(0) qm_free(0x401ff000, 0x403b62a8), called from t_stats.c: free_tm_stats(227) 0(0) qm_free: freeing frag. 0x403b6290 alloc'ed from t_stats.c: init_tm_stats(171) 0(0) qm_free(0x401ff000, 0x403b6250), called from t_stats.c: free_tm_stats(228) 0(0) qm_free: freeing frag. 0x403b6238 alloc'ed from t_stats.c: init_tm_stats(163) 0(0) DEBUG: tm_shutdown : done 0(0) qm_free(0x80fbe60, 0x81304bc), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x81304a4 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x40235bfc), called from sl_stats.c: sl_stats_destroy(164) 0(0) qm_free: freeing frag. 0x40235be4 alloc'ed from sl_stats.c: init_sl_stats(179) 0(0) qm_free(0x401ff000, 0x4023603c), called from sl_funcs.c: sl_shutdown(99) 0(0) qm_free: freeing frag. 0x40236024 alloc'ed from sl_funcs.c: sl_startup(82) 0(0) qm_free(0x80fbe60, 0x81303cc), called from sr_module.c: destroy_modules(357) 0(0) qm_free: freeing frag. 0x81303b4 alloc'ed from sr_module.c: register_module(136) 0(0) qm_free(0x401ff000, 0x40233474), called from tcp_main.c: destroy_tcp(1367) 0(0) qm_free: freeing frag. 0x4023345c alloc'ed from tcp_main.c: init_tcp(1303) 0(0) qm_free(0x401ff000, 0x402334dc), called from tcp_main.c: destroy_tcp(1371) 0(0) qm_free: freeing frag. 0x402334c4 alloc'ed from tcp_main.c: init_tcp(1326) 0(0) qm_free(0x401ff000, 0x4023450c), called from tcp_main.c: destroy_tcp(1375) 0(0) qm_free: freeing frag. 0x402344f4 alloc'ed from tcp_main.c: init_tcp(1338) 0(0) qm_free(0x401ff000, 0x402334a8), called from tcp_main.c: destroy_tcp(1380) 0(0) qm_free: freeing frag. 0x40233490 alloc'ed from tcp_main.c: init_tcp(1315) 0(0) qm_free(0x401ff000, 0x40233440), called from timer.c: destroy_timer(80) 0(0) qm_free: freeing frag. 0x40233428 alloc'ed from timer.c: init_timer(57) 0(0) qm_free(0x80fbe60, 0x8134104), called from timer.c: destroy_timer(89) 0(0) qm_free: freeing frag. 0x81340ec alloc'ed from timer.c: register_timer(104) 0(0) qm_free(0x80fbe60, 0x8133fbc), called from timer.c: destroy_timer(89) 0(0) qm_free: freeing frag. 0x8133fa4 alloc'ed from timer.c: register_timer(104) 0(0) qm_free(0x80fbe60, 0x81344cc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81344b4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813448c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134474 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813444c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134434 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813440c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81343f4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x81343cc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x81343b4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813438c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134374 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813434c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8134334 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8134044), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x813402c alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133f7c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133f64 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133f3c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133f24 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133efc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133ee4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133ebc), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133ea4 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x8133d3c), called from unixsock_server.c: close_unixsock_server(652) 0(0) qm_free: freeing frag. 0x8133d24 alloc'ed from unixsock_server.c: unixsock_register_cmd(673) 0(0) qm_free(0x80fbe60, 0x813430c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81342f4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81342cc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81342b4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813428c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134274 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813424c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134234 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813420c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81341f4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81341cc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x81341b4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x813418c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8134174 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8134004), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133fec alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133e7c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133e64 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133e3c), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133e24 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133dfc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133de4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133dbc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133da4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x8133cfc), called from fifo_server.c: destroy_fifo(158) 0(0) qm_free: freeing frag. 0x8133ce4 alloc'ed from fifo_server.c: register_fifo_cmd(133) 0(0) qm_free(0x80fbe60, 0x81340c4), called from script_cb.c: destroy_script_cb(81) 0(0) qm_free: freeing frag. 0x81340ac alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x80fbe60, 0x8133d7c), called from script_cb.c: destroy_script_cb(81) 0(0) qm_free: freeing frag. 0x8133d64 alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x80fbe60, 0x8134084), called from script_cb.c: destroy_script_cb(83) 0(0) qm_free: freeing frag. 0x813406c alloc'ed from script_cb.c: register_script_cb(53) 0(0) qm_free(0x401ff000, 0x4023553c), called from main.c: cleanup(373) 0(0) qm_free: freeing frag. 0x40235524 alloc'ed from main.c: main(1534) 0(2180) shm_mem_destroy 0(2180) destroying the shared memory lock The ser.cfg that I have used is the same included in the compilation except for the debugs to get activated. Next I include it in case you see something rare: Any help or idea is very very welcome :-) Thank you very much Jesús ********************************************************************** # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (src_ip==193.175.135.0/24){ force_send_socket(smaug:5080); forward(193.175.135.179); break; } # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

1 0

[Serusers] Ser+acc+dbtext
by Xavier TRENTIN 10 Jun '05

10 Jun '05

Hi all, I've ser running, and I want to log calls into a dbtext database. I wrotte an empty table with the same fields as in the acc table generated for mysql. When ser is running, I can see that the acc table loaded in memory is updated, but not my acc table file. The file is updated only once ser is stopped. I'd like to know if it's possible to put the data into the acc table file while ser is running, and how to do that? Thanks. Best Regards. Xavier.

3 3

[Serusers] mass user creation
by Gerd Feiner 10 Jun '05

10 Jun '05

Hi, what would be a suggested way of creating huge batches of users at once? serctl is obviously not usable for that and simple insertion in in the database would miss too much of those IDs which serctl creates, right? Also, this user-creation should create RPIDs and aliases as well as URI-entries. Brgds, Gerd

2 1

[Serusers] Is manipulating the content of an AVP possible?
by Kanakatti Mahesh Subramanya 10 Jun '05

10 Jun '05

to be precise a) how do I go about combining two AVPs ? b) how can I change the value (string manip, to be precise) the value of an AVP? c) how can I do (b), based on the content of a different AVP? Ideally, the answer isn't "write the code to do this" The <hack>method</hack> that I came up with was a) write the AVPs I care about to a custom header b) resend the request to myself, and then use 'subst' (or equivalent textops command) needless to say, urghk. anybody? cheers

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[Serusers] acc empty timestamp
by TCCHAN 10 Jun '05

10 Jun '05

Dear All, I am using 0.8.14 and I managed to get acc modules to write to MySQL, but it seems that the timestamp field is blank. Would the acc module fill the timestamp or did I missed something? Thank for any help available. Regards, TC Chan

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Re: [Serusers] SER with pa module including publish (ERROR)
by Jesús M. Movilla 10 Jun '05

10 Jun '05

Than you for your suggestions. I wil try it and I will tell you. Now I am trying to install the 0.9.0 version which also includes a publish.c class in the pa module ;-). Maybe, i will be luckier :-). But I found problems when starting with mysql.so module because this module was not compiled when executed "make bin" (I removed the mysql from the exclude_modules). Anyway I will tell you later. So kind for your answers Hope I can help you too in the future Regards Jesús ----- Original Message ----- From: "Samuel Osorio Calvo" <samuel.osorio(a)nl.thalesgroup.com> To: <jlbravo(a)acotelsa.com>; <jamey.hicks(a)hp.com>; <jmms03(a)tid.es> Cc: <serusers(a)lists.iptel.org> Sent: Friday, June 10, 2005 10:36 AM Subject: Re: [Serusers] SER with pa module including publish (ERROR) Have you checked the format of the config file?? It might be in DOS format and SER is not able to load it. Just to be sure you can use dos2unix command (I had some problem with some version in the sense that this command modified the permissions of the file upon converting....) Hope it helps, Samuel. Unclassified. >>> "Jesús M. Movilla" <jmms03(a)tid.es> 06/10/05 08:22AM >>> Hi again, I tried to startup SER deleting the first line of ser.cfg but it didn't work (similar problem). However I installed the 0.8.14 SER version and there was no problem in starting up the SER with the previous failing ser.cfg. ¿any other idea? Thanks for your answers Jesús P.D.- The ser.cf used is included down here. I put comments on the proper function (handle_subscription, for example) of 0.10. pa module version to start it up properly with 0.8.14 version. #################################################################### # global configuration parameters a debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #user=ser #group=ser #fifo_user=ser # owner of the ser fifo #fifo_group=ser #fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- pa params -- # #modparam("pa", "default_expires", 3600) #Set default_expires parameter #modparam("pa", "default_priority_percentage", 50) #Set default_priority_percentage parameter #modparam("pa", "db_url", "mysql:ser:heslo@localhost/ser") #Set db_url parameter #modparam("pa", "pa_domain", "hi.inet") #Set pa_domain parameter # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; #if (!uri=~"hi.inet" { # mark routing logic in request # append_hf("P-hint: outbound\r\n"); # route(1); # break; #}; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"hi.inet") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("hi.inet", "subscriber")) { www_challenge("hi.inet", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri=~"hi.inet") { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; #handle_publish("registrar"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] running ser+freeradius with mysql
by gerard＠bubus.com 10 Jun '05

10 Jun '05

Hi list, I have a big problem with this. I have ser_head working with freeradius 0.9.1 and radiusclint-ng. Everything looks like working, ser start doesn´t give me any problem, I have create a database with ser_mysql.sh create, and I have configured a softphone X-Lite to check it all. But when it makes the register, radius doesn´t look like accessing to MySQL. Please any help will be genial!!! Gerard Sanz. This is the ser response: sip:~ # ser start Listening on udp: sip.mars.sol [192.168.10.2]:5060 tcp: sip.mars.sol [192.168.10.2]:5060 Aliases: *: mars.sol:* WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 111616 0(0) INFO: udp_init: SO_RCVBUF is finally 223232 2(0) INFO: fifo process starting: 4630 2(4630) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 0(4625) REGISTER sip:sip.mars.sol From:sip:laptop@sip.mars.sol To:sip:laptop@sip.mars.sol 0(4625) REGISTER: Authenticating user 0(4625) REGISTER: challenging user This is the radius response: rad_recv: Access-Request packet from host 127.0.0.1:32854, id=99, length=195 User-Name = "laptop(a)mars.sol" Digest-Attributes = "\n\010laptop" Digest-Attributes = "\001\nmars.sol" Digest-Attributes = "\002*42adc7dfc79504c6f4aae461eb995f8889b49658" Digest-Attributes = "\004\022sip:sip.mars.sol" Digest-Attributes = "\003\nREGISTER" Digest-Response = "4fcfd64bf5b67211395a084e4c978eba" Service-Type = Sip-Session Sip-Uri-User = "laptop" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 rlm_eap: EAP-Message not found rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "laptop" Digest-Realm = "mars.sol" Digest-Nonce = "42adc7dfc79504c6f4aae461eb995f8889b49658" Digest-URI = "sip:sip.mars.sol" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST rlm_sql (sql): Reserving sql socket id: 1 rlm_sql_mysql: MYSQL check_error: 1146 received rlm_sql_getvpdata: database query error rlm_sql (sql): SQL query error; rejecting user rlm_sql (sql): Released sql socket id: 1 this is my sql.conf from the radius server: # # Configuration for the SQL module, when using MySQL. # # The database schema is available at: # # src/radiusd/src/modules/rlm_sql/drivers/rlm_sql_mysql/db_mysql.sql # # If you are using PostgreSQL, please use 'postgresql.conf', instead. # If you are using Oracle, please use 'oracle.conf', instead. # If you are using MS-SQL, please use 'mssql.conf', instead. # # $Id: sql.conf,v 1.26.4.1 2003/08/26 12:26:57 phampson Exp $ # sql { # Database type # Current supported are: rlm_sql_mysql, rlm_sql_postgresql, # rlm_sql_iodbc, rlm_sql_oracle, rlm_sql_unixodbc, rlm_sql_freetds driver = "rlm_sql_mysql" # Connect info server = "localhost" login = "ser" password = "heslo" # Database table configuration radius_db = "ser" and finally, this is my ser.cfg: # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode #debug=7 #fork=no #log_stderror=yes */ listen=sip.mars.sol # IP address(es) or hostnames where SER will listen for messages port=5060 # IP port where SER will listen for messages alias=mars.sol # additional IP address(es) or hostnames to satisfy the condition "uri==myself" check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/ser_radius/lib/ser/modules/mysql.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/sl.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/tm.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/rr.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/usrloc.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/registrar.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/textops.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/uri.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/ser_radius/lib/ser/modules/auth.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/auth_db.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/ser_radius/lib/ser/modules/xlog.so" #loadmodule "/usr/local/ser_radius/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- acc params -- #modparam("acc", "log_level", 1) #modparam("acc", "radius_flag", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_INFO","%rm %ru From:%fu To:%tu\n"); # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # loose-route processing (grant Route routing if route headers present) if (loose_route()) { xlog("L_INFO"," -->Request loose_routed\n"); append_hf("P-hint: loose_routed by biloxi-proxy \r\n"); t_relay(); break; }; # record route if its a session (INVITE method) if (method=="INVITE") { record_route(); }; #check if request-URI is an alias lookup ("aliases"); #if request URI is numeric and starting with 1 forward to sip.mars.sol if (uri=~"^sip:1[0-9]*") { forward("sip.mars.sol"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { log(1, "REGISTER: Authenticating user\n"); # Uncomment this if you want to use digest authentication if (!radius_www_authorize("mars.sol")) { log(1, "REGISTER: challenging user\n"); www_challenge("mars.sol", "0"); break; }; save("location"); break; }; if (method=="INVITE") { log(1, "INVITE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="MESSAGE") { log(1, "MESSAGE\n"); setflag(1); /* set for accounting (the same value as in log_flag!) */ }; if (method=="BYE" || method=="CANCEL") { log (1, "BYE or CANCEL\n"); setflag(1); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Thank you very much. I hope somebody has had the same problem before and could solve it. Gerard

2 1

Re: [Serusers] SER with pa module including publish (ERROR)
by Samuel Osorio Calvo 10 Jun '05

10 Jun '05

Have you checked the format of the config file?? It might be in DOS format and SER is not able to load it. Just to be sure you can use dos2unix command (I had some problem with some version in the sense that this command modified the permissions of the file upon converting....) Hope it helps, Samuel. Unclassified. >>> "Jesús M. Movilla" <jmms03(a)tid.es> 06/10/05 08:22AM >>> Hi again, I tried to startup SER deleting the first line of ser.cfg but it didn't work (similar problem). However I installed the 0.8.14 SER version and there was no problem in starting up the SER with the previous failing ser.cfg. ¿any other idea? Thanks for your answers Jesús P.D.- The ser.cf used is included down here. I put comments on the proper function (handle_subscription, for example) of 0.10. pa module version to start it up properly with 0.8.14 version. #################################################################### # global configuration parameters a debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode */ fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #user=ser #group=ser #fifo_user=ser # owner of the ser fifo #fifo_group=ser #fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) # # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- pa params -- # #modparam("pa", "default_expires", 3600) #Set default_expires parameter #modparam("pa", "default_priority_percentage", 50) #Set default_priority_percentage parameter #modparam("pa", "db_url", "mysql:ser:heslo@localhost/ser") #Set db_url parameter #modparam("pa", "pa_domain", "hi.inet") #Set pa_domain parameter # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; #if (!uri=~"hi.inet" { # mark routing logic in request # append_hf("P-hint: outbound\r\n"); # route(1); # break; #}; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~"hi.inet") { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("hi.inet", "subscriber")) { www_challenge("hi.inet", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri=~"hi.inet") { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (method=="PUBLISH") { if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); }; #handle_publish("registrar"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] ser AAA with freeradius and mysql backend example
by Gerard 10 Jun '05

10 Jun '05

Dear Jan, I've tried but the freeradius doen't look like connecting ok with the mySQL. This is the message it gives me when I try auth.: rad_recv: Access-Request packet from host 127.0.0.1:32854, id=99, length=195 User-Name = "laptop(a)mars.sol" Digest-Attributes = "\n\010laptop" Digest-Attributes = "\001\nmars.sol" Digest-Attributes = "\002*42adc7dfc79504c6f4aae461eb995f8889b49658" Digest-Attributes = "\004\022sip:sip.mars.sol" Digest-Attributes = "\003\nREGISTER" Digest-Response = "4fcfd64bf5b67211395a084e4c978eba" Service-Type = Sip-Session Sip-Uri-User = "laptop" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 rlm_eap: EAP-Message not found rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "laptop" Digest-Realm = "mars.sol" Digest-Nonce = "42adc7dfc79504c6f4aae461eb995f8889b49658" Digest-URI = "sip:sip.mars.sol" Digest-Method = "REGISTER" rlm_digest: Adding Auth-Type = DIGEST rlm_sql (sql): Reserving sql socket id: 1 rlm_sql_mysql: MYSQL check_error: 1146 received rlm_sql_getvpdata: database query error rlm_sql (sql): SQL query error; rejecting user rlm_sql (sql): Released sql socket id: 1 Have a nice day, gerard(a)bubus.com

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[Serusers] SER with pa module including publish (ERROR)
by Jesús M. Movilla 10 Jun '05

10 Jun '05

Hi to all, I would like to know the way to install SER with PUBLISH support in the pa module. I tried to compile the pa module from CVS version and inserted within the 0.8.14 version but it didn´t work:I obtained the following error when starting up: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: /usr/local/lib/ser/modules/pa.so: undefined symbol qm_free I also tried compiling the whole CVS version (0.10.99) but when I start up I obtain the ERROR bad config file (1 errors). I also tried with the config file ser.cfg from the 0.8.14 version but I obtained the same error. So, could someone tell me how to proceed in order to have a SER server with a pa module implementing REGISTER, SUSBSCRIBE AND PUBLISH. Thank you very much Jesús M. Movilla

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[Serusers] Record-Route and Route parameters patch
by Bogdan-Andrei Iancu 10 Jun '05

10 Jun '05

We are all aware of SER limitation due the fact it is *only* transaction statefull at proxy level, any dialog support being missing. There are many features/services which actually requires a dialog awareness, like storing the information in the dialog creation stage, information which will be used during the whole dialog existence. The most urging example is NAT traversal, in dealing with the within the dialog INVITEs (re-INVITEs). When processing the initial INVITE, the proxy detects if the caller or callee is behind some NAT and fixes the signalling and media parts - since not all the detection mechanism are available for within the dialog requests (like usrloc), to be able to fix correspondingly the sequential requests, the proxy must remember that the original request was NAT processed. There are many other cases where dialog awareness fixes or helps....it's not my goal to make a list here.... The solution is to store additional dialog-related information in the routing set, headers which show up in all sequential requests; by routing set I would say the Route hdrs and RURI; the routing set is build based on Record-Route and Contact hdrs. So far, the way to go was using the Contact header (via parameters) as carrier. In most of the cases works, but this mechanism is not RFC reinforced - I had a talk with Juha and he found some phones which actually ignore the Contact hdr parameters, which is perfect allowed by RFC....Not to mention that which Contact hdr (UAC or UAS) appear in routing info depends on the request direction (UAC->UAS or UAS->UAC) and Record-Route parameters are re-infonced by RFC (see 12.1.1 UAS behavior) So the next try is to add this information in the Record-Route header inserted by the proxy. Within the dialog, this information can be found (with no direction dependencies) in Route header (with or proxy address)... And here is a patch for the RR module which adds this functionality: 1) adding parameters to Record-Route; since RR is inserted during req. processing (as lump), no textops function can be apply to it. So use the add_rr_param("param") function - it can be called also before or after the record_route() call; I would say it's more convenient from scripting point of view (no dependencies between doing record_route() and adding params); 2) checking Route parameters: this could be done via textops, but it would be dangerous -> you have to identify the Route header corresponding to your proxy (which will be quite difficult to put as regexp), So use check_route_param("regexp") function which will check the proper Route parameters against the given regexp. Must be call after loose_route(). Basic scenario (example): UAC SER PROXY UAS ------ INVITE ----------> record_route() --------- INVITE --------> add_rr_param(";foo=true") ----- reINVITE ---------> loose_route() -------- reINVITE -------> check_route_param(";foo=true") <---- reINVITE ---------- loose_route() <------- reINVITE -------- check_route_param(";foo=true") <-------- BYE ----------- loose_route() <--------- BYE ----------- check_route_param(";foo=true") The patch doesn't alter any of the previous functionality. Jan take a look and let's put it on CVS. bogdan Index: modules/rr/loose.c =================================================================== RCS file: /cvsroot/ser/sip_router/modules/rr/loose.c,v retrieving revision 1.36 diff -u -r1.36 loose.c --- modules/rr/loose.c 23 Feb 2005 17:16:05 -0000 1.36 +++ modules/rr/loose.c 15 Apr 2005 14:52:55 -0000 @@ -30,6 +30,8 @@ * --------- * 2003-02-28 scratchpad compatibility abandoned (jiri) * 2003-01-27 next baby-step to removing ZT - PRESERVE_ZT (jiri) + * 2005-04-10 check_route_param() and all hooks for keeping reference to + * Route params added (bogdan) */ @@ -57,6 +59,10 @@ #define ROUTE_SUFFIX ">\r\n" #define ROUTE_SUFFIX_LEN (sizeof(ROUTE_SUFFIX)-1) +/* variables used to hook the param part of the local route -bogdan */ +static unsigned int routed_msg_id; +static str routed_params = {0,0}; + /* * Test whether we are processing pre-loaded route set @@ -511,12 +517,16 @@ if (is_myself(&puri.host, puri.port_no)) #endif { - /* if (enable_double_rr && is_2rr(&_ruri->params)) { */ - /* DBG("ras(): Removing 2nd URI of mine: '%.*s'\n", rt->nameaddr.uri.len, ZSW(rt->nameaddr.uri.s)); */ + /*if (enable_double_rr && is_2rr(&_ruri->params)) { + * DBG("ras(): Removing 2nd URI of mine: '%.*s'\n", + * rt->nameaddr.uri.len, ZSW(rt->nameaddr.uri.s)); */ + /* set the hooks for the params -bogdan */ + routed_msg_id = _m->id; + routed_params = puri.params; if (!rt->next) { - /* No next route in the same header, remove the whole header - * field immediately - */ + /* No next route in the same header, remove the whole header + * field immediately + */ if (!del_lump(_m, hdr->name.s - _m->buf, hdr->len, 0)) { LOG(L_ERR, "after_strict: Cannot remove Route HF\n"); return RR_ERROR; @@ -661,11 +671,15 @@ if (is_myself(&puri.host, puri.port_no)) #endif { - DBG("after_loose: Topmost route URI: '%.*s' is me\n", uri->len, ZSW(uri->s)); + DBG("after_loose: Topmost route URI: '%.*s' is me\n", + uri->len, ZSW(uri->s)); + /* set the hooks for the params -bogdan */ + routed_msg_id = _m->id; + routed_params = puri.params; if (!rt->next) { - /* No next route in the same header, remove the whole header - * field immediately - */ + /* No next route in the same header, remove the whole header + * field immediately + */ if (!del_lump(_m, hdr->name.s - _m->buf, hdr->len, 0)) { LOG(L_ERR, "after_loose: Can't remove Route HF\n"); return RR_ERROR; @@ -684,16 +698,16 @@ if (enable_double_rr && is_2rr(&puri.params)) { if (!rt->next) { - /* No next route in the same header, remove the whole header - * field immediately - */ + /* No next route in the same header, remove the whole header + * field immediately + */ if (!del_lump(_m, hdr->name.s - _m->buf, hdr->len, 0)) { LOG(L_ERR, "after_loose: Can't remove Route HF\n"); return RR_ERROR; } res = find_next_route(_m, &hdr); if (res < 0) { - LOG(L_ERR, "after_loose: Error while finding next route\n"); + LOG(L_ERR,"after_loose: Error while finding next route\n"); return RR_ERROR; } if (res > 0) { /* No next route found */ @@ -783,3 +797,32 @@ } } } + + + +int check_route_param(struct sip_msg * msg, char *re, char *foo) +{ + regmatch_t pmatch; + char bk; + + /* check if the hooked params belong to the same message */ + if (routed_msg_id != msg->id) + return -1; + + /* check if params are present */ + if ( !routed_params.s || !routed_params.len ) + return -1; + + /* do the well-known trick to convert to null terminted */ + bk = routed_params.s[routed_params.len]; + routed_params.s[routed_params.len] = 0; + DBG("DEBUG:rr:check_route_param: params are <%s>\n", routed_params.s); + if (regexec( (regex_t*)re, routed_params.s, 1, &pmatch, 0)!=0) { + routed_params.s[routed_params.len] = bk; + return -1; + } else { + routed_params.s[routed_params.len] = bk; + return 1; + } +} + Index: modules/rr/loose.h =================================================================== RCS file: /cvsroot/ser/sip_router/modules/rr/loose.h,v retrieving revision 1.3 diff -u -r1.3 loose.h --- modules/rr/loose.h 24 Aug 2004 09:00:38 -0000 1.3 +++ modules/rr/loose.h 15 Apr 2005 14:52:56 -0000 @@ -40,4 +40,6 @@ int loose_route(struct sip_msg* _m, char* _s1, char* _s2); +int check_route_param(struct sip_msg * msg, char *re, char *foo); + #endif /* LOOSE_H */ Index: modules/rr/record.c =================================================================== RCS file: /cvsroot/ser/sip_router/modules/rr/record.c,v retrieving revision 1.14 diff -u -r1.14 record.c --- modules/rr/record.c 24 Aug 2004 09:00:38 -0000 1.14 +++ modules/rr/record.c 15 Apr 2005 14:52:56 -0000 @@ -29,6 +29,7 @@ * History: * ------- * 2003-04-04 Extracted from common.[ch] (janakj) + * 2005-04-10 add_rr_param() function and all corresponing hooks added (bogdan) */ #include <string.h> @@ -63,6 +64,22 @@ #define INBOUND 1 /* Insert inbound Record-Route */ #define OUTBOUND 0 /* Insert outbound Record-Route */ +#define RR_PARAM_BUF_SIZE 512 + +static unsigned int last_rr_msg; + +/* RR suffix lump and offset - used for inserting RR params + * after RR was done; use a two values array for INBOUND and + * OUTBOUND RR -bogdan */ +static struct lump *rr_suffix_lump[2] = {0,0}; +static int rr_suffix_end_offset[2] = {0,0}; + +/* RR param buffer - usd for storing RR param which are + * added before RR insertion -bogdan */ +static char rr_param_buf_ptr[RR_PARAM_BUF_SIZE]; +static str rr_param_buf = {rr_param_buf_ptr,0}; +static unsigned int rr_param_msg; + /* * Extract username from the Request URI @@ -104,16 +121,20 @@ { char* prefix, *suffix, *crlf, *r2; int suffix_len, prefix_len; + char *p; prefix_len = RR_PREFIX_LEN + (user->len ? (user->len + 1) : 0); prefix = pkg_malloc(prefix_len); if (enable_full_lr) { suffix_len = (_lr ? RR_LR_FULL_TERM_LEN : RR_SR_TERM_LEN) + - ((tag && tag->len) ? (RR_FROMTAG_LEN + tag->len) : 0); + ((tag && tag->len) ? (RR_FROMTAG_LEN + tag->len) : 0) + + rr_param_buf.len; } else { suffix_len = (_lr ? RR_LR_TERM_LEN : RR_SR_TERM_LEN) + - ((tag && tag->len) ? (RR_FROMTAG_LEN + tag->len) : 0); + ((tag && tag->len) ? (RR_FROMTAG_LEN + tag->len) : 0) + + rr_param_buf.len; } + rr_suffix_end_offset[_inbound] = RR_SR_TERM_LEN; suffix = pkg_malloc(suffix_len); crlf = pkg_malloc(2); @@ -145,21 +166,24 @@ #endif prefix[RR_PREFIX_LEN + user->len] = '@'; } - + + p = suffix; if (tag && tag->len) { - memcpy(suffix, RR_FROMTAG, RR_FROMTAG_LEN); - memcpy(suffix + RR_FROMTAG_LEN, tag->s, tag->len); - if (enable_full_lr) { - memcpy(suffix + RR_FROMTAG_LEN + tag->len, _lr ? RR_LR_FULL_TERM : RR_SR_TERM, _lr ? RR_LR_FULL_TERM_LEN : RR_SR_TERM_LEN); - } else { - memcpy(suffix + RR_FROMTAG_LEN + tag->len, _lr ? RR_LR_TERM : RR_SR_TERM, _lr ? RR_LR_TERM_LEN : RR_SR_TERM_LEN); - } + memcpy(p, RR_FROMTAG, RR_FROMTAG_LEN); + p += RR_FROMTAG_LEN; + memcpy(p, tag->s, tag->len); + p += tag->len; + } + if (rr_param_buf.len) { + memcpy( p, rr_param_buf.s, rr_param_buf.len); + p += rr_param_buf.len; + } + if (enable_full_lr) { + memcpy( p, _lr ? RR_LR_FULL_TERM : RR_SR_TERM, + _lr ? RR_LR_FULL_TERM_LEN : RR_SR_TERM_LEN); } else { - if (enable_full_lr) { - memcpy(suffix, _lr ? RR_LR_FULL_TERM : RR_SR_TERM, _lr ? RR_LR_FULL_TERM_LEN : RR_SR_TERM_LEN); - } else { - memcpy(suffix, _lr ? RR_LR_TERM : RR_SR_TERM, _lr ? RR_LR_TERM_LEN : RR_SR_TERM_LEN); - } + memcpy( p, _lr ? RR_LR_TERM : RR_SR_TERM, + _lr ? RR_LR_TERM_LEN : RR_SR_TERM_LEN); } memcpy(crlf, CRLF, 2); @@ -177,6 +201,7 @@ r2 = 0; } if (!(_l2 = insert_new_lump_before(_l2, suffix, suffix_len, 0))) goto lump_err; + rr_suffix_lump[_inbound] = _l2; suffix = 0; if (!(_l2 = insert_new_lump_before(_l2, crlf, 2, 0))) goto lump_err; crlf = 0; @@ -225,6 +250,12 @@ tag = 0; } + if (rr_param_buf.len && rr_param_msg!=_m->id) { + /* rr_params were set for a different message -> reset buffer */ + rr_param_buf.len = 0; + DBG("----> reset rr_param_buf\n"); + } + if (enable_double_rr) { l = anchor_lump(_m, _m->headers->name.s - _m->buf, 0, 0); l2 = anchor_lump(_m, _m->headers->name.s - _m->buf, 0, 0); @@ -256,6 +287,8 @@ return -4; } + /* reset the rr_param buffer */ + rr_param_buf.len = 0; return 0; } @@ -266,8 +299,6 @@ */ static inline int do_RR(struct sip_msg* _m, int _lr) { - static unsigned int last_rr_msg; - if (_m->id == last_rr_msg) { LOG(L_ERR, "record_route(): Double attempt to record-route\n"); return -1; @@ -395,3 +426,72 @@ { return do_RR(_m, 0); } + + +/* + * Appends a string to an existent lump + */ +static inline int append_to_lump( struct lump *l, str *s, int end_offset) +{ + char *p; + + /* update buffer */ + p = pkg_realloc( l->u.value, l->len+s->len); + if (p==0) { + LOG(L_ERR,"ERROR:rr:add_rr_param: no more pkg memory\n"); + return -1; + } + memmove(p+l->len-end_offset+s->len, p+l->len-end_offset, end_offset); + memcpy( p+l->len-end_offset, s->s, s->len); + /* update lump structure */ + l->len += s->len; + l->u.value = p; + return 0; +} + + +/* + * Appends a new Record-Route parameter + */ +int add_rr_param(struct sip_msg* msg, char* param, char* foo) +{ + str *rr_param; + int i; + + rr_param = (str*)param; + + if (last_rr_msg==msg->id) { + /* RR was already done -> have to modify the RR-sufix lump */ + for( i=OUTBOUND ; i<=INBOUND ; i++ ) { + if ( !rr_suffix_lump[i] || !rr_suffix_end_offset[i] ) + continue; + if (append_to_lump( rr_suffix_lump[i], rr_param, + rr_suffix_end_offset[i])!=0) { + LOG(L_ERR,"ERROR:rr:add_rr_param: failed to update lump\n"); + goto error; + } + } + } else { + /* RR not done yet -> store the param in the static buffer */ + if (rr_param_msg!=msg->id) { + /* it's about a different messge -> reset buffer */ + rr_param_buf.len = 0; + rr_param_msg = msg->id; + } + if (rr_param_buf.len+rr_param->len>RR_PARAM_BUF_SIZE) { + LOG(L_ERR,"ERROR:rr:add_rr_param: maximum size of " + "rr_param_buf exceeded\n"); + goto error; + } + memcpy( rr_param_buf.s+rr_param_buf.len, rr_param->s, rr_param->len); + rr_param_buf.len += rr_param->len; + DBG("DEBUG:rr:add_rr_param: rr_param_buf=<%.*s>\n",rr_param_buf.len, + rr_param_buf.s); + } + return 1; + +error: + return -1; +} + + Index: modules/rr/record.h =================================================================== RCS file: /cvsroot/ser/sip_router/modules/rr/record.h,v retrieving revision 1.2 diff -u -r1.2 record.h --- modules/rr/record.h 24 Aug 2004 09:00:38 -0000 1.2 +++ modules/rr/record.h 15 Apr 2005 14:52:56 -0000 @@ -56,4 +56,10 @@ int record_route_strict(struct sip_msg* _m, char* _s1, char* _s2); +/* + * Appends a new Record-Route parameter + */ +int add_rr_param(struct sip_msg* msg, char* param, char* foo); + + #endif /* RECORD_H */ Index: modules/rr/rr_mod.c =================================================================== RCS file: /cvsroot/ser/sip_router/modules/rr/rr_mod.c,v retrieving revision 1.33 diff -u -r1.33 rr_mod.c --- modules/rr/rr_mod.c 24 Aug 2004 09:00:38 -0000 1.33 +++ modules/rr/rr_mod.c 15 Apr 2005 14:52:57 -0000 @@ -33,11 +33,14 @@ * 2003-03-19 all mallocs/frees replaced w/ pkg_malloc/pkg_free (andrei) * 2003-04-01 Added record_route with ip address parameter (janakj) * 2003-04-14 enable_full_lr parameter introduced (janakj) + * 2005-04-10 add_rr_param() and check_route_param() added (bogdan) */ #include <stdio.h> #include <stdlib.h> +#include <regex.h> + #include "../../sr_module.h" #include "../../ut.h" #include "../../error.h" @@ -61,6 +64,7 @@ static int mod_init(void); static int str_fixup(void** param, int param_no); +static int regexp_fixup(void** param, int param_no); /* @@ -74,10 +78,18 @@ * Oh, BTW, have I mentioned already that you shouldn't use strict routing ? */ static cmd_export_t cmds[] = { - {"loose_route", loose_route, 0, 0, REQUEST_ROUTE}, - {"record_route", record_route, 0, 0, REQUEST_ROUTE}, - {"record_route_preset", record_route_preset, 1, str_fixup, REQUEST_ROUTE}, - {"record_route_strict" , record_route_strict, 0, 0, 0 }, + {"loose_route", loose_route, 0, 0, + REQUEST_ROUTE}, + {"record_route", record_route, 0, 0, + REQUEST_ROUTE}, + {"record_route_preset", record_route_preset, 1, str_fixup, + REQUEST_ROUTE}, + {"record_route_strict" , record_route_strict, 0, 0, + 0}, + {"add_rr_param", add_rr_param, 1, str_fixup, + REQUEST_ROUTE}, + {"check_route_param", check_route_param, 1, regexp_fixup, + REQUEST_ROUTE}, {0, 0, 0, 0, 0} }; @@ -149,3 +161,29 @@ return 0; } + + +static int regexp_fixup(void** param, int param_no) +{ + regex_t* re; + + if (param_no==1) { + if ((re=pkg_malloc(sizeof(regex_t)))==0) { + LOG(L_ERR,"ERROR:rr:regexp_fixup: no more pkg memory\n"); + return E_OUT_OF_MEM; + } + if (regcomp(re, *param, REG_EXTENDED|REG_ICASE|REG_NEWLINE) ) { + pkg_free(re); + LOG(L_ERR, "ERROR:rr:regexp_fixup: bad regexp %s\n",(char*)*param); + return E_BAD_RE; + } + /* free string */ + pkg_free(*param); + /* replace it with the compiled re */ + *param=re; + } + return 0; +} + + +

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[Serusers] sems voicemail: t_relay / t_newtran and socket closing problem
by Dennis Real 09 Jun '05

09 Jun '05

Hi, I have a strange problem with stateful processing using t_relay and t_newtran. For normal routing I use t_relay but all sems modules require the usage of t_newtran, e.g.: if ( !t_newtran() ) { sl_reply_error(); }; t_write_unix("/tmp/am_sock", "voicemail"); The problem now is how to handle BYE and CANCEL messages for deestablishing connections. For the echo server (reachable at 4242(a)192.168.11.1) I use the following code for getting back transaction and closing socket to sems: if ( uri=~"^sip:4242@" && ( method=="BYE" || method=="CANCEL" ) ) { if (!t_newtran()) { sl_send_reply("500", "could not create transaction"); break; }; if( !t_write_unix("/tmp/am_sock", "bye") ) { t_reply("500", "error contacting sems"); }; break; } This works fine. But how can I do this for voicemail? The uri will not be changed so I can't destinguish between BYE for a normal and a voicemail connection. All other destinations do not use t_newtran (and no socket to sems) so simply t_relay is used for BYE and CANCEL messages. The whole ser.cfg is attached below. Regards and thanks for your help, Dennis # ser.cfg # ----------- global configuration parameters ------------------------ debug=3 fork=yes log_stderror=yes check_via=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode=0666 fifo_db_url="mysql://ser:heslo@localhost/ser" listen=192.168.11.1 listen=127.0.0.1 alias=192.168.11.1 alias=127.0.0.1 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/speeddial.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 1) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1 ) modparam("acc", "log_missed_flag", 2) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "log_fmt", "cdfimorstup") modparam("xlog", "buf_size", 8192) modparam("registrar", "default_expires", 180) modparam("registrar", "nat_flag", 3) modparam("mediaproxy", "natping_interval", 52) modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetrics-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") # ------------------------- request routing logic ------------------- # # main routing logic # # -------------------------------------------------------------------- route { if ( !mf_process_maxfwd_header("10") ) { log(1, "LOG: Too many hops. LOOP?\n"); if ( method!="ACK" ) { sl_send_reply("483", "Too many hops"); } break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if ( uri=~"^sip:4242@" && ( method=="BYE" || method=="CANCEL" ) ) { if (!t_newtran()) { sl_send_reply("500", "could not create transaction"); break; }; if( !t_write_unix("/tmp/am_sock", "bye") ) { t_reply("500", "error contacting sems"); }; break; } else if ( method=="BYE" || method=="CANCEL" ) { end_media_session(); setflag(1); } if ( client_nat_test("3") && !search("^Record-Route:") ) { force_rport(); fix_contact(); }; if ( loose_route() ) { if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; t_relay(); break; }; if ( method=="ACK" ) { use_media_proxy(); } ###### if ( !(uri==myself) ) { append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if ( !t_relay() ) { sl_reply_error(); break; }; break; }; ###### if ( uri==myself ) { if ( method=="REGISTER" ) { if ( client_nat_test("3") ) { setflag(3); force_rport(); fix_contact(); } if ( !www_authorize("192.168.11.1", "subscriber") ) { www_challenge("192.168.11.1", "1"); break; } if ( !save("location") ) { sl_reply_error(); break; }; break; } # method==register else if ( method=="INVITE" ) { if ( !proxy_authorize("", "subscriber") ) { proxy_challenge("", "1" ); break; }; t_on_failure("1"); t_on_reply("1"); use_media_proxy(); # ------------------------------------------------------------------------ # echo server # ------------------------------------------------------------------------ if ( uri=~"^sip:4242@.*" ) { # switch to statefull mode: if ( !t_newtran() ) { sl_send_reply("500","could not create transaction"); break; }; if( !t_write_unix("/tmp/am_sock", "echo") ) { t_reply("500", "error contacting sems"); }; break; } # offline? if ( !lookup("location") ) { if ( !t_newtran() ) { sl_reply_error(); }; t_write_unix("/tmp/am_sock", "voicemail"); break; }; } # method==invite # route append_hf("P-hint: USRLOC\r\n"); if ( !t_relay() ) { end_media_session(); sl_reply_error(); break; }; } ; # uri==myself } /* end of initial routing logic */ ############################################################################################ onreply_route[1] { if ( status=~"(183)|(2[0-9][0-9])" ) { if ( client_nat_test("1") ) { fix_contact(); }; use_media_proxy(); }; } failure_route[1] { end_media_session(); }

3 2

Re: [Serdev] Re: [Serusers] how to increment cseq? (for uac auth)
by Michael Ulitskiy 09 Jun '05

09 Jun '05

On Thursday 09 June 2005 05:07 am, you wrote: > Just my 2 cents.... > > Since SER is transaction aware and not dialog aware, it is not easy to maintain consistent Cseq values during a dialog. I'm sure UAC maintainers are thinking on it and they will come with a solution, but just let them work without stress ;) Sure. I didn't want to put stress on anyone. However I proposed a way to implement it by existing (or simple to add) means. Not that I liked it too much myself, but that is the only thing I could come with and I'd really like to receive some comments on whether it's possible or I'm completely wrong. Also I'm wondering how hard is it to have something like "save_dialog()" and "check_dialog()" function that would store callid and from/to tag in some in-memory hash. I don't think that performance/memory consumption objections really stands anymore. How much memory would it take to save dialog identifier (callid and tags)? Let's say 100 bytes. So if I dedicate 10M of RAM for that table I'd be able to keep 100000 dialogs. > However, don't think the UAC approach is the right one having TLS around......I would prefer pushing for TLS instead of adding a table of existing dialog in SER with the appropriate values. I'm sure there's lots of providers reading this list so if everybody starts asking for TLS....well, big providers will start providing it for auth. I'm not an expert but I've been told implementing TLS is not a hard job....;) Well, if TLS becomes mainstream I'll have no choice, but use it, but I'd prefer to do without TLS or even moving to TCP, at least for a while. I believe existing md5 authentication is good enough for what it is used in SIP. Sure setting TLS is doable, but it's definitely more work then just setting shared password. Also I don't think TLS is going to become mainstream in SIP any time soon. Look at TLS in SMTP. It's been available for years, but I still don't see it in wide use for server-to-server communication, just between servers and end users. > > Samuel. > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 06/09/05 07:18AM >>> > > Michael Ulitskiy wrote: > > Greg, do you really believe that avoiding added, but optional > > complexity > > is good enough reason for not having IMHO necessary functionality? > > Not at all. My point was that it is extremely important to add functionality > where it belongs logically and where you can ensure that regular ser.cfg > users are able to use the functionality without understanding the specs. > > > Again, IMHO, having something better than ip auth is a must in > > today's internet. > > Again, I agree. > > > On the subject: > > I guess the reason is, as I said before, that not only cseq in > > original invite must be incremented, but cseq in all subsequent > > in-dialog messages must > > be adjusted (decremented for messages relayed to callee and > > incremented > > for messages relayed to called party). That is what, I guess, is hard > > to do in today's ser. > > I've posted my thoughts on how it possibly could be accomplished > > at http://lists.iptel.org/pipermail/serdev/2005-May/004589.html and > > haven't received any feedback or comments. > > I guess either there's a lack of interest for getting it to work > > which I guess is really strange or it's too hard to implement at > > current stage. > > As always any comments are welcome :) > > I don't think it's a lack of interest or lack of need for it. All the > developers are bombarded with requests and have long lists of core things to > do. Sometimes its even hard to get them to evaluate and include patches in > CVS. > IMHO, increment cseq belongs in the UAC module (can do far too much harm > if users play around with it). If you cannot get the maintainers attention, > there is only one way: Get your hands dirty ;-) > > g-) > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > >

2 1

[Serusers] how to increment cseq? (for uac auth)
by Stefan Sayer 09 Jun '05

09 Jun '05

Hello, is there a simple method to increment CSeq of a branch? I have the following in failure_route: if (uac_auth()) { # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("3"); # repeat the request with auth response append_branch(); ----------> and here I'd like to increment CSeq t_relay(); } If CSeq is not incremented, this authenticated request does not work with RFC strict UAs, as discussed eg in this (http://lists.iptel.org/pipermail/serusers/2005-May/019806.html) thread. Now as I want to proxy auth an INVITE I am sending myself from sems I could do a dirty workaround: assume that the INVITE needs two CSeqs and increment cseq a second time for subsequent requests in sems. This is of course a hack but would save me from implementing auth in sems for the moment. Thanks for any suggestions Stefan

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[Serusers] Question about configuring Calls Made tab on SerWeb
by Cameron Black 09 Jun '05

09 Jun '05

Hi, on the calls made page it shows less actual results than the number shown in the page links. For example, on page 1 it says showing 1-20 of 83 but if i go through each page, from page 4 onwards they are blank. Had a quick check with the query and it was returning 50 calls made. Any ideas? Also, the calls made seems to be made up of the same calls but broken down into segments - ie there are 4 rows for what seems like 1 call. Cam. -- -- |Cameron Black Broadband Solutions for |Application Developer Home & Business @ |PlusNet plc. www.plus.net +------ PlusNet - The smarter way to Internet! -----

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[Serusers] Authentication problem.
by c waddy 09 Jun '05

09 Jun '05

I can call SER with a phone registered to it, no problem. When i try to make outbound calls to my provider, they fail with. 407 - proxy authentication required. I cant seem to trace it down: if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("sr.mydomain.com", "subscriber")) { www_challenge("sr.mydomain", "0"); break; }; save("aliases"); save("location"); break; }; Is this a problem at my end or my providers? I am new to SER, is there something i am missing? Thanks.

2 1

Re: [Serusers] How to configure pa?
by zhaomin 09 Jun '05

09 Jun '05

I have same problem in xmpp messages , I can not understand well those module's configure (jabber.so ,pa.so),Can I used the jid as a ser account? Thanks Zhaomin ----- Original Message ----- From: "Steven Poveda" <stepoca(a)hotmail.com> To: <serusers(a)iptel.org> Sent: Wednesday, May 25, 2005 6:42 AM Subject: [Serusers] How to configure pa? > Hi, > > I have just installed SER and I can see the server is not supporting PUBLISH method. I have loaded pa module but the reject to PUBLISH method is stil there. > > Where can I find information regarding pa configuration ? > > Regards, > > Steven > > > > > -------------------------------------------------------------------------------- > Don't just search. Find. MSN Search Check out the new MSN Search! -------------------------------------------------------------------------------- > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] How to configure pa?
by Steven Poveda 09 Jun '05

09 Jun '05

2 1

[Serusers] Re: SEMS not working, problem solved, thank you
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

[Serusers] ATA phone not working
by Min ZOU 09 Jun '05

09 Jun '05

Hi, I am trying to get an ATA phone working for SER server. Now the ATA phone can receive calls from softphone(kphone) on a PC, while it cannot call out. (softphone, ATA phone and SER server are in the same subnet) $ serctl ul show 100 <sip:100@192.168.1.100:5600>;q=0.00;expires=802 Any idea on how to resolve the problem? Thanks! -- Best regards, Min

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[Serusers] add_rcv_param() for INVITEs
by Martin Koenig 09 Jun '05

09 Jun '05

Hello, is it possible to use add_rcv_param() also for Contact-HF of INVITEs, instead of fix_nated_contact()? This would make sure that the NATed End Device gets fed exactly the contact as R-URI that it announced in the INVITE: Example R-URI in Loose-Route Requests for fix_nated_contact(): Ser (1.2.3.4): INVITE device(a)4.5.6.7 Route: 1.2.3.4;lr=on Device (10.0.0.1, NAT 4.5.6.7): INVITE device(a)123.123.123.123 With add_rcv_param(): Ser: INVITE device(a)10.0.0.1;rcv_param=4.5.6.7 Route: 1.2.3.4;lr=on Device: INVITE device(a)10.0.0.1 The problem is that the End-Device, in the fix_nated_contact() callflow, has to accept a Request that is not really for this device (IP in Request-URI != IP of the Device). Imagine a Ser checkip uri == myself. This would fail without knowing the public IP of the NAT and hard-conding it into Ser.cfg. Does Ser support the rcv_param in Request-URI also? What do you think? Any experience on this? Regards, Martin

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[Serusers] SER, help please!
by Ryan Pagquil 09 Jun '05

09 Jun '05

Hi, Whenever I uncomment i change the integer value of modparam("usrloc", "db_mode", 0) changed to 1 or 2 and when i about to restart SER using ./serctl stop then ./serctl start, SER strat but when i check ps aux | grep ser no SER is running. I checked /var/log/debug/ and i got this messages: Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : start Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : empting DELETE list Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : empting hash table Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown: releasing timers Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : removing semaphores Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : done Jun 9 00:45:10 sip ./ser[7553]: shm_mem_destroy Jun 9 00:45:10 sip ./ser[7553]: destroying the shared memory lock Whats the meaning of destroying the shared memory lock? I'm running SER on FreeBSD/sparc64. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 1

[Serusers] Re: [Sems] SEMS not working
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

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Re: [Serdev] Re: [Serusers] how to increment cseq? (for uac auth)
by Samuel Osorio Calvo 09 Jun '05

09 Jun '05

Just my 2 cents.... Since SER is transaction aware and not dialog aware, it is not easy to maintain consistent Cseq values during a dialog. I'm sure UAC maintainers are thinking on it and they will come with a solution, but just let them work without stress ;) However, don't think the UAC approach is the right one having TLS around......I would prefer pushing for TLS instead of adding a table of existing dialog in SER with the appropriate values. I'm sure there's lots of providers reading this list so if everybody starts asking for TLS....well, big providers will start providing it for auth. I'm not an expert but I've been told implementing TLS is not a hard job....;) Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/09/05 07:18AM >>> Michael Ulitskiy wrote: > Greg, do you really believe that avoiding added, but optional > complexity > is good enough reason for not having IMHO necessary functionality? Not at all. My point was that it is extremely important to add functionality where it belongs logically and where you can ensure that regular ser.cfg users are able to use the functionality without understanding the specs. > Again, IMHO, having something better than ip auth is a must in > today's internet. Again, I agree. > On the subject: > I guess the reason is, as I said before, that not only cseq in > original invite must be incremented, but cseq in all subsequent > in-dialog messages must > be adjusted (decremented for messages relayed to callee and > incremented > for messages relayed to called party). That is what, I guess, is hard > to do in today's ser. > I've posted my thoughts on how it possibly could be accomplished > at http://lists.iptel.org/pipermail/serdev/2005-May/004589.html and > haven't received any feedback or comments. > I guess either there's a lack of interest for getting it to work > which I guess is really strange or it's too hard to implement at > current stage. > As always any comments are welcome :) I don't think it's a lack of interest or lack of need for it. All the developers are bombarded with requests and have long lists of core things to do. Sometimes its even hard to get them to evaluate and include patches in CVS. IMHO, increment cseq belongs in the UAC module (can do far too much harm if users play around with it). If you cannot get the maintainers attention, there is only one way: Get your hands dirty ;-) g-) _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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[Serusers] Re: [Sems] SEMS not working
by Samuel Osorio Calvo 09 Jun '05

09 Jun '05

First of all: there's no need to send the e-mail more than once...it will just make people not answering. Probably what happens is that you have the old libraries (the ones you compiled with the HEAD version of SEMS) and SEMS rel_0.8.12. Be sure the libraries are correct. Samuel Unclassified. >>> Llanos Serna García-Conde <llanosserna(a)hotmail.com> 06/09/05 09:49AM >>> Hi all, I'm trying to make SEMS work. First I was working with SER 0.8.14 and SEMS cvs but I always got the error consume_request(): consumed. I only want SEMS for voicemail, so the list recommended that I downloaded the SEMS rel_0.8.12, which is supposed to be compatible with SER 0.8.14, but when loading the plug-ins I get errors about undefined symbols. I've been reading previous posts and it seems that it's either an incompatibility between versions, or that SEMS is working with another version's modules or something like that. I don't know how to solve this. Here are the errors I'm getting: Configuration: configuration file: /usr/local/etc/sems/sems12.conf Ser's FIFO: /tmp/ser_fifo our FIFO: /tmp/am_fifo plug-in path: /usr/local/src/sems_rel_0_8_12/lib daemon mode: 0 local IP: 172.25.97.19 (4851) DEBUG: main (sems.cpp:386): Loading application plug-ins (4851) DEBUG: load (AmPlugIn.cpp:89): loading /usr/local/src/sems_rel_0_8_12/lib/apps/bot.so ... (4851) ERROR: loadPlugIn (AmPlugIn.cpp:103): AmPlugIn::loadPlugIn: /usr/local/src/sems_rel_0_8_12/lib/apps/bot.so: undefined symbol: _ZN12AmRequestUAC7dialoutERKSsS1_S1_S1_ (4851) ERROR: load (AmPlugIn.cpp:91): while loading plug-in '/usr/local/src/sems_rel_0_8_12/lib/apps/bot.so' (4850) WARNING: sig_usr_fifo (sems.cpp:84): signal 17 received (4850) INFO: sig_usr_fifo (sems.cpp:93): finished Please help, thanks a lot, Llanos

1 0

[Serusers] fifo issues with ser cvs head version
by Cameron Beattie 09 Jun '05

09 Jun '05

I have installed the cvs head version of ser as of today. Ser seems to start OK but serctl moni reports: Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config I am using the same ser.cfg file as was running successfully on v 0.9.0. Could anyone advise what I'm doing wrong? ser.cfg debug=3 fork=yes log_stderror=yes dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pword@localhost/ser" sock_mode=0666 listen=ip.ip.ip.ip port=5060 children=4 Regards Cameron

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[Serusers] parse_hfs(): Error while parsing headers
by mma 09 Jun '05

09 Jun '05

i 'm connecting to SER, which running on 192.168.1.70 'sip.server.net', using Windows Messenger i add user2 to my contact list i have this errors in response to a SUBSCRIBE request parse_hfs(): Error while parsing headers handle_subscription(): Error while parsing message header can somebody help me Regards //////////////////////////////////////////////////////////////////////////////// 0(2945) SIP Request: 0(2945) method: <REGISTER> 0(2945) uri: <sip:sip.server.net> 0(2945) version: <SIP/2.0> 0(2945) parse_headers: flags=2 0(2945) end of header reached, state=5 0(2945) parse_headers: Via found, flags=2 0(2945) parse_headers: this is the first via 0(2945) After parse_msg... 0(2945) preparing to run routing scripts... 0(2945) parse_headers: flags=100 0(2945) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2945) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2945) parse_headers: flags=200 0(2945) end of header reached, state=9 0(2945) DEBUG: get_hdr_field: <To> [34]; uri=[sip:user1@sip.server.net] 0(2945) DEBUG: to body [<sip:user1@sip.server.net>^M ] 0(2945) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(2945) DEBUG: get_hdr_body : content_length=0 0(2945) found end of header 0(2945) find_first_route: No Route headers found 0(2945) loose_route: There is no Route HF 0(2945) grep_sock_info - checking if host==us: 18==18 && [sip.server.net] == [sip.server.net] 0(2945) grep_sock_info - checking if port 5060 matches port 5060 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) parse_headers: flags=4000000 0(2945) build_contact(): Created Contact HF: Contact: <sip:192.168.1.118:14310>;expires=3600^M 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) check_via_address(192.168.1.118, 192.168.1.118, 0) 0(2945) DEBUG:destroy_avp_list: destroying list (nil) 0(2945) receive_msg: cleaning up 0(2945) SIP Request: 0(2945) method: <SUBSCRIBE> 0(2945) uri: <sip:user2@sip.server.net> 0(2945) version: <SIP/2.0> 0(2945) parse_headers: flags=2 0(2945) end of header reached, state=5 0(2945) parse_headers: Via found, flags=2 0(2945) parse_headers: this is the first via 0(2945) After parse_msg... 0(2945) preparing to run routing scripts... 0(2945) parse_headers: flags=100 0(2945) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2945) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2945) parse_headers: flags=10 0(2945) DEBUG: add_param: tag=963f17727b404c5ca92bd1ad97491e13 0(2945) DEBUG: add_param: epid=db4638b430 0(2945) end of header reached, state=29 0(2945) parse_headers: flags=200 0(2945) end of header reached, state=9 0(2945) DEBUG: get_hdr_field: <To> [33]; uri=[sip:user2@sip.server.net] 0(2945) DEBUG: to body [<sip:user2@sip.server.net>^M ] 0(2945) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 0(2945) DEBUG: get_hdr_body : content_length=0 0(2945) found end of header 0(2945) find_first_route: No Route headers found 0(2945) loose_route: There is no Route HF 0(2945) grep_sock_info - checking if host==us: 18==18 && [sip.server.net] == [sip.server.net] 0(2945) grep_sock_info - checking if port 5060 matches port 5060 0(2945) DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) parse_headers: flags=78 0(2945) t_lookup_request: start searching: hash=53801, isACK=0 0(2945) DEBUG: proceeding to pre-RFC3261 transaction matching 0(2945) DEBUG: t_lookup_request: no transaction found 0(2945) handle_subscription() entered 0(2945) parse_headers: flags=304010 0(2945) parse_hfs(): Error while parsing headers 0(2945) handle_subscription(): Error while parsing message header 0(2945) handle_subscription about to send_reply and return -2 0(2945) DEBUG: t_check: msg id=2 global id=2 T start=0x422efda0 0(2945) DEBUG: t_check: T already found! 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) check_via_address(192.168.1.118, 192.168.1.118, 0) 0(2945) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(2945) DEBUG: add_to_tail_of_timer[2]: 0x422efde8 0(2945) DEBUG: reply sent out. buf=0x8138d64: SIP/2.0 4..., shmem=0x422f13d8: SIP/2.0 4 0(2945) DEBUG: _reply_light: finished 0(2945) DEBUG:destroy_avp_list: destroying list (nil) 0(2945) receive_msg: cleaning up 0(2945) SIP Request:

1 1

[Serusers] Problem in downloading ser-0.9.0_src.tar.gz
by Gaurav Kansal 09 Jun '05

09 Jun '05

Hi On trying to download ser-0.9.0_src.tar.gz from <ftp://ftp.berlios.de/pub/ser/0.9.0/src/ser-0.9.0_src.tar.gz> ftp://ftp.berlios.de/pub/ser/0.9.0/src/ser-0.9.0_src.tar.gz, i get a pop up window for username/password. In that I give the username/password with which I am registered at iptel.org. But I get an error message and download fails. Can anybody tell me how to download it. Regards, Gaurav Kansal

2 1

[Serusers] SEMS not working
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

re:[Serusers] two PSTN gateways with SER
by Ahmad Cheikh Moussa 09 Jun '05

09 Jun '05

Hi! I found this thread in the mail archive and I have a question to this thread. How many pstn gateways can I handle with this failover statement ? Is there a timeout I have to keep in mind, or can I put gateways as many as I have ??? For example. I have one SER Server and 10 PSTN gateways and on every gateway, 4 PRIs. Can I configure the ser server so that the server ask one after another until he found a gateway which accept his call ? The last question is, which isdn features can I use with SER ? I mean can a Ser user make auto hold or call park or something like that ? Thanks in advance, Ahmad >You could not, because ser does not know which PSTN gateway is less >loaded. What you can do is to forward all calls to one of the gateways >and when the gateway refuses the call then you can try the other one. >This can be done using failure_route statement in the configuration >file. > > Jan. > >On 27-05 15:34, Gustavo García Bernardo wrote: >> Hi, >> >> I have a platform based in SER SIP proxy, and i would like to forward >PSTN >> calls to two Cisco gateways. How could implement the logic for send >the >> calls to the less loaded gateway? >> >> Best regards. >> >> G. -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

3 5

[Serusers] Ser Presence Server + Windows MSN 4.7
by Simith Nambiar 09 Jun '05

09 Jun '05

Hello Serusers, I'am new to the List , please can you give me the below details , 1.) Does the Presence server in Ser do SUBSCRIBE/ NOTIFY processing and is it compatible with Windows Messenger 4.7 ? what i mean to say is that , does it reflect the current status of the logged in users, and generate appropriate Notifications ? Please let me know ! Thank You, Cheers. Simith

1 0

[Serusers] Ser Presence Server and Windows Messenger
by Simith Nambiar 09 Jun '05

09 Jun '05

Hi All, Does the Presence server in Ser do SUBSCRIBE/ NOTIFY processing and is it compatible with Windows Messenger 4.7 ? what i mean to say is that , does it reflect the current status of the logged in users, and generate appropriate Notifications ? Thank You, Cheers. Simith

1 0

[Serusers] SerWeb ul_add: flags expected
by Mr Greg Plater 09 Jun '05

09 Jun '05

I found the error was an extra field in the table subscriber that was not in pending.. causing mismatch. Pays to check everything when you start to customize serweb.

1 0

[Serusers] SerWeb ul_add: flags expected
by Mr Greg Plater 09 Jun '05

09 Jun '05

Hi When I register a new user I get an error after clicking on the confirmation link in the email I receive..NOTE not before. Also when I have a user in serweb and I try to add an Alias I see this in red: 400 ul_add: flags expected in debug I see : 1(1880) ERROR: ul_add: flags expected This was all working until I decided to make a better ser.cfg file. I think maybe an issue with ser.cfg and needing something for aliases. Because with the old ser.cfg users could register with out issue and I could create an alias. Perms has been added to subscriber and pending tables This has been added to config.php var $flags; $this->flags="128";

1 0

[Serusers] Sems plugin:bot.so, undefined symbol
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

[Serusers] optimizing mysql queries
by Iqbal 08 Jun '05

08 Jun '05

Hi The two queries below being used to pull acc info out, seem very slow, are there any Mysql whizzes on the list, the db is optimized, but the second query is just far too long for me to debug. select count(*) from acc t1 right outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t1.sip_method='INVITE' where t2.sip_method='BYE' and isnull(t1.username) and (t2.username='040633' and t2.domain='sip.domain.com') and t2.callee_deleted != '1' (select t1.to_uri as inv_to_uri, t1.sip_to as inv_sip_to,t1.sip_callid as inv_callid, t1.time as inv_time, t1.fromtag as inv_fromtag, t1.sip_status as inv_status, t2.to_uri as bye_to_uri,t2.sip_to as bye_sip_to, t2.sip_callid as bye_callid,t2.time as bye_time,t2.fromtag as bye_fromtag,t2.totag as bye_totag,t2.from_uri as bye_from_uri,t2.sip_from as bye_sip_from,sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) as length,ifnull(t1.time, t2.time) as ttime , 'outgoing' as call_type from acc t1 left outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t2.sip_method='BYE' where t1.sip_method='INVITE' and (t1.username='040633' and t1.domain='sip.domain.com') and t1.caller_deleted != '1' ) union (select t1.to_uri as inv_to_uri, t1.sip_to as inv_sip_to,t1.sip_callid as inv_callid,t1.time as inv_time, t1.fromtag as inv_fromtag,t1.sip_status as inv_status,t2.to_uri as bye_to_uri,t2.sip_to as bye_sip_to,t2.sip_callid as bye_callid, t2.time as bye_time,t2.fromtag as bye_fromtag,t2.totag as bye_totag,t2.from_uri as bye_from_uri, t2.sip_from as bye_sip_from,sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) as length, ifnull(t1.time, t2.time) as ttime , 'outgoing' as call_type from acc t1 right outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t1.sip_method='INVITE' where t2.sip_method='BYE' and isnull(t1.username) and (t2.username='040633' and t2.domain='sip.domain.com') and t2.callee_deleted != '1' )order by ttime desc limit 0, 20 Now I did a "explain" on these (after having added indexes, since the default install doesn't add indexes. +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ | table | type | possible_keys | key | key_len | ref | rows | Extra | +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ | t1 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | | | t2 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t1 | ref | sip_method | sip_method | 129 | const | 2782 | Using where; Not exists | | t1 | index | NULL | username | 64 | NULL | 6837 | Using index | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | Using where; Using index | +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ this for the second query, so rough maths tells me that 351554245281666581856 rows must be examined in order to return the data set..which is alot of rows, in any language. the explain for the first query returns +-------+------+---------------------+------------+---------+-------+------+-------------+ | table | type | possible_keys | key | key_len | ref | rows | Extra | +-------+------+---------------------+------------+---------+-------+------+-------------+ | t1 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | | +-------+------+---------------------+------------+---------+-------+------+-------------+ which is about 4million. Iqbal

1 1

[Serusers] job opening in NYC
by Igor Balk 08 Jun '05

08 Jun '05

Senior Systems Engineer StanaPhone Communications, LLC is one of the leading Voice over Internet Protocol (VoIP) service providers and is a subsidiary of Intermedia.NET. We are opening a search for a Senior Systems Engineer. The position is located in our headquarters in New York City. With industry leaders as founders and a dynamic and growing team, this is a great opportunity to join an early-stage company with a bright future. If you're an enthusiastic and creative individual with more then 3 years of experience then this is a great career opportunity for you. You must be able to work in an very fast-paced and dynamic environment. As an early stage team member you'll have a unique opportunity to experience an open and collaborative environment in which you'll be able to help shape company strategy and direction. Responsibilities: You will work as a member of the network team, designing and developing next generation VoIP services. You must have strong knowledge of new and existing telephony protocols and their internetworking and a forward-looking perspective with the vision to develop new, innovative services with evolving protocols. Desired Skills: * Demonstrated expertise in the following areas: SIP, ENUM, TLS, TRIP, SS7, development in a Linux / UNIX environment, C and Perl programming, network programming (TCP/UDP) * Working knowledge of the following software systems: Asterisk PBX; SIP Express Router or other SIP Proxy Servers, MySQL and Oracle databases. * Cisco certification is a definite plus. * Strong troubleshooting, debugging, problem solving, and analytical skills * Effective communication (verbal and written) and interpersonal skills * You should be authorized to work in the US as we do not sponsor visas at this time. Intermedia.NET, the parent company, is a leading provider of advanced Web and Exchange Hosting Solutions. The company has been a pioneer in hosting services since 1995 and has continued to innovate, providing its customers with HostPilot, an advanced, industry-recognized control panel for server management and administration. The company responds to customers' most essential needs by providing cost effective, secure and reliable hosting solutions to more than 15,000 businesses. Headquartered in New York City, with offices and resellers worldwide, Intermedia.NET continues to be a customer-focused, financially stable and dependable company. Intermedia.NET is a Microsoft Certified Partner. More information can be found at: www.intermedia.net. Intermedia.NET is an equal opportunity employer and offers competitive wages. We provide top benefits and compensation. Please email resumes with wage history and requirements to jobs(a)stanaphone.com with "Systems Engineer" in the subject line. Please respect our time - no calls or walk-ins.

1 0

[Serusers] ser is failing after a while with sems
by Roger Lewau 08 Jun '05

08 Jun '05

I have set up ser with mediaproxy and sems according to example configs. When I start up mediaproxy, ser and sems all phones register properly and calls can be placed internally (have no external gate yet). I also tried calling a UA not onlie to get sems to take the call and it works great. So I tell my beta tester to start testing.... 1 day after, all calls are rejected because of server error. I have to restart everything and it works again, only to fail the day after... These entries are found in the log for every call I try to make when ser stopped working. Jun 8 22:42:33 ns2 ser[523]: ERROR: t_newtran: transaction already in process 0x29830c58 Jun 8 22:42:33 ns2 ser[523]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Error message on SJPhone is "number not available" The error message about t_newtran tells me this is caused by a part of the config added for sems to work. Also it seems that after putting the t_newtran in the config there should be some t_release as well since I get this message repeatedly after restarting ser. However I can not find any documentation stating this is necessary or where this should be put in place within ser.cfg. Jun 8 22:53:34 ns2 ser[4898]: WARNING: script writer didn't release transaction If I comment out the SEMS configuration lines withing ser.cfg and restart ser. It works properly again.... First, is there any any one with a ser.cfg script that works with sems without the error/waring messages above being logged? If so please send it to me. Secondly, have any one written any official documentation about this product, including different scenarios? Since the system is widely used I do not understand why there is no official documentation available. Dont say onsip, no more help to find there.

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[Serusers] REGISTER, auth & INVITE
by Lol Zimmerli 08 Jun '05

08 Jun '05

Hello list! I do have something like this in my ser.cfg: if (method=="REGISTER") { if (!www_authorize("domain.com", "subscriber")) { www_challenge("domain.com", "1"); break; }; break; }; It works perfectly. And if I set a line in my snom with inexistant account, of course REGISTER fails: [2]8/6/2005 16:29:39: Registrar 9898(a)212.xxx.xxx.xxx refused with code 401 But, with the same un-REGISTERed phone, I can phone: INVITE works :( I then tried something like this: if (method=="INVITE"){ if (src_ip != "212.xxx.xxx.xxx"){ if (!www_authorize("domain.com", "subscriber")) { www_challenge("domain.com", "1"); break; }; }; else{ route(1); }; }; And then, SER fail to restart: Starting SIP Express Router ERROR: bad config file (6 errors) startproc: exit status of parent of /usr/local/sbin/ser: 255 Is my approach wrong or am I doing a little typo-like error? Best regards -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Write clearly - don't sacrifice clarity for "efficiency". - The Elements of Programming Style (Kernighan & Plaugher)

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[Serusers] Socketser command interface
by Rodrigo P. Telles 08 Jun '05

08 Jun '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, We did a socket interface for serctl command, it means you can run serctl command in your SER box remotely. It's primitive yeat only accept a few commands. It's wrote using Perl Socket interface. Few free to use (at your own risk), change and redistribute! You can download the server and a simple client to test your server from: http://www.devel-it.org/socketser Please read socketser code before use it. Instructions to install, run and use included! We've been using it for 4 months without problems I hope you enjoy! Best regards. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCpgz1iLK8unYgEMQRAqRvAJ4nTOaHzCM9H01YTrqXLE+wJ35hEQCeJUWZ +br5k43W1OkYdMRScJiidcQ= =qJvW -----END PGP SIGNATURE-----

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RE: [Serusers] Re-routing to same device in failure_route
by Karun Chemudugunta 08 Jun '05

08 Jun '05

Yes, you are right. Both INVITE messages are same except branch id. Any ideas on how can we solve this issue ? Thanks, karun -----Original Message----- From: ggb(a)tid.es [mailto:ggb@tid.es] Sent: Tuesday, June 07, 2005 3:29 PM To: Karun Chemudugunta; serusers(a)lists.iptel.org Cc: serdev(a)lists.iptel.org Subject: RE: [Serusers] Re-routing to same device in failure_route Hi, It's possible that the Xten softphone detects that the second INVITE is part of a cleared session (the first INVITE rejected with 400 error), and discards it.... G. _____ De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de Karun Chemudugunta Enviado el: martes, 07 de junio de 2005 18:43 Para: serusers(a)lists.iptel.org CC: serdev(a)lists.iptel.org Asunto: [Serusers] Re-routing to same device in failure_route Hi Group, Cisco 5300 -----> SER ----> Xten softphone-A ----> Failed call with error code >400 -------> try to route to Xten softphone A(same as previous) I am trying to route calls to softphones in such way that, if call failed by some reason (>400), trying to route second time to the same soft phone using append_brach() and t_relay() like below. But, soft phone is unable process INVITE message for some reason. But, if I forward failed call different soft phone, call was successful. Am I doing any thing wrong in config ??? Thanks for your help in advance. Regards, Karun ########################################################################################## # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #alias=voip.cafe.bevocal.com #alias=66.77.14.238 alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { log(1,"Inside failure_route five"); revert_uri(); append_branch(); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); }; } ######################################################################################

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Re: [Serusers] UAC module: multiple PSTN gateways
by Nour Omar 08 Jun '05

08 Jun '05

Some providers use the IP of the request for authentication and billing. I was wondering how SER can handle that?? Marian Dumitru <marian.dumitru(a)voice-sistem.ro> wrote:Hi Mohammad, UAC module is not present at all in 0.9.0 version, *only* in CVS head version! So far only the CSeq problem is know to us. What exactly doesn't work for you? regards, Marian info(a)beeplove.com wrote: > I tried to configure uac module using this example on ser 0.9.0 > didn't have good luck :-( > > I am just curious to know, > Are you using cvs version of ser to make uac worked? > So far I know, there have a cseq issue. Is it solved yet? > > Thanks > MOhammad > > > Original Message: > ----------------- > From: Marian Dumitru marian.dumitru(a)voice-sistem.ro > Date: Wed, 08 Jun 2005 16:42:24 +0300 > To: info(a)beeplove.com, ext_news(a)appsfarm.com, serusers(a)lists.iptel.org > Subject: Re: [Serusers] UAC module: multiple PSTN gateways > > > Hi Mohammad, > > have a look at an example: > http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth > > the UAC module is present only in CVS head. > > regards, > Marian > > Mohammad Khan wrote: > >>Did you guys was able to use UAC module successfully? >>If yes, would you please tell what version of ser and uac module you are >>using? >> >>Thanks, >>MOhammad >> >> >>Marian Dumitru wrote: >> >> >>>Hi Cameron, >>> >>>You can specify multiple authentication credentials by just >>>enumerating several "credential" parameters for the module: >>> modparam("uac","credential","user1:realm1:passwd1") >>> modparam("uac","credential","user2:realm2:passwd2") >>> modparam("uac","credential","user3:realm3:passwd3") >>> >>>The module will choose the correct credentials to use based on the >>>"realm" from the auth challenge. >>> >>>regards, >>>Marian >>> >>> >>>Cameron Beattie wrote: >>> >>> >>>>Is it possible to specify multiple authentication credentials within >>>>the UAC module? What I want to do is access multiple PSTN gateways >>>>that each require different credentials. The basic purpose of this is >>>>to allow least cost routing. >>>> >>>>Example >>>>Provider 1 offers cheap calling to +44 (UK) >>>>Provider 2 offers cheap calling to +1 (US) >>>>When user dials a 44 destination ser should relay to Provider 1 >>>>gateway, which will require authentication using Credential 1. >>>>When user dials a 1 destination ser should relay to Provider 2 >>>>gateway, which will require authentication using Credential 2. >>>> >>>>Any advice would be greatfully received. >>>> >>>>Regards >>>> >>>>Cameron >>> >>> >>> >> > -- Voice System http://www.voice-system.ro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] UAC module: multiple PSTN gateways
by info＠beeplove.com 08 Jun '05

08 Jun '05

I tried to configure uac module using this example on ser 0.9.0 didn't have good luck :-( I am just curious to know, Are you using cvs version of ser to make uac worked? So far I know, there have a cseq issue. Is it solved yet? Thanks MOhammad Original Message: ----------------- From: Marian Dumitru marian.dumitru(a)voice-sistem.ro Date: Wed, 08 Jun 2005 16:42:24 +0300 To: info(a)beeplove.com, ext_news(a)appsfarm.com, serusers(a)iptel.org Subject: Re: [Serusers] UAC module: multiple PSTN gateways Hi Mohammad, have a look at an example: http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth the UAC module is present only in CVS head. regards, Marian Mohammad Khan wrote: > Did you guys was able to use UAC module successfully? > If yes, would you please tell what version of ser and uac module you are > using? > > Thanks, > MOhammad > > > Marian Dumitru wrote: > >> Hi Cameron, >> >> You can specify multiple authentication credentials by just >> enumerating several "credential" parameters for the module: >> modparam("uac","credential","user1:realm1:passwd1") >> modparam("uac","credential","user2:realm2:passwd2") >> modparam("uac","credential","user3:realm3:passwd3") >> >> The module will choose the correct credentials to use based on the >> "realm" from the auth challenge. >> >> regards, >> Marian >> >> >> Cameron Beattie wrote: >> >>> Is it possible to specify multiple authentication credentials within >>> the UAC module? What I want to do is access multiple PSTN gateways >>> that each require different credentials. The basic purpose of this is >>> to allow least cost routing. >>> >>> Example >>> Provider 1 offers cheap calling to +44 (UK) >>> Provider 2 offers cheap calling to +1 (US) >>> When user dials a 44 destination ser should relay to Provider 1 >>> gateway, which will require authentication using Credential 1. >>> When user dials a 1 destination ser should relay to Provider 2 >>> gateway, which will require authentication using Credential 2. >>> >>> Any advice would be greatfully received. >>> >>> Regards >>> >>> Cameron >> >> >> > > -- Voice System http://www.voice-system.ro -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] ser+pstn call restrict
by Szabolcs Szasz 08 Jun '05

08 Jun '05

Hi! I am new in ser configuration. My configuration: I use Audiocodes Mediant 2000 for PSTN gateway, ser-0.8.14 installed on my redhat 9.0 box, audiocodes and welltech FXS's as user agents. Now, I can call succesfully sip-->sip, sip-->PSTN and PSTN-->sip. My problem is that I need to restrict some calls for some users, international, national and local (sip-->sip calls). Can anyone help me? Here is my ser.cfg file. Thanks! # # ----------- global configuration parameters ------------------------ debug=4 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" listen=10.1.10.10 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0)# add value to ;lr param to make some broken UAs happy # -- rr params -- modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # forward all calls to audiocodes with 0 as the digit header if ( (uri=~"sip:0[0-9]*@*") | (uri=~"sip:1111@*") ){ rewritehostport("10.1.10.1:5060"); forward(10.1.10.1, 5060); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("10.1.10.10", "subscriber")) { www_challenge("10.1.10.10", "0"); break; }; save("location"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } _____________________________ Szabolcs Szasz Digi Communication SRL H2/3 Petofi Sandor St. Sovata, Mures, Romania sszasz(a)digicomm.ro www.digicomm.ro _____________________________

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[Serusers] Re: [Serdev] ENUM issues - req. for discussion#
by Samuel Osorio Calvo 08 Jun '05

08 Jun '05

Then I see it would take too much effort to add an asynchronous resolver, but I don't think the lack of memory is the right answer: your proposal of using 500 children is more memory demanding than forking, isn't it? You can just add a configuration variable, max_fork=500 and with , for instance, children=3 your deployment will require much less memory than just children=500!!!!!!!!! Nevertheless, for simplicity (in the sense of less points of failure and a more stable system) it is better to stay with the "common" resolver. Samuel. Unclassified. >>> Andrei Pelinescu-Onciul <andrei(a)iptel.org> 06/07/05 04:49PM >>> On Jun 07, 2005 at 14:56, Samuel Osorio Calvo <samuel.osorio@nl.thalesgroup..com> wrote: > >If DNS is slow, or misconfigured (e.g. a zone is delegated to a > >nameserver which is down), the thread will be blocked for several > >seconds. E.g. if you use debian woody and 2 nameservers in > >/etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS > >allows configuration of the DNS timeouts. Nevertheless, you have to > >consider that a ser thread will be blocked up to 20 seconds. This has > >impacts on your configuration: > > I don't know the details but would it be really difficult to use an asyncrhonous resolver, such as resiprocate SIP stack does with ARES?? Besides exec_* calls, the main SER's performance bottelneck is the DNS resolving step thus it would be a great improvement adding asyncronous DNS queries. Using asynchronous dns would work as long as you have memory to save the state of the pending dns request. It could be easily attacked in the same way (lots of DNS requests that will take a long time to resolve => out of memory => no more messages processed). Besides using it would mean saving the complete state of the message and of the ser processing of the message in the moment the dns request was made. For example if you make a dns request in module foo, function bar() you should be able to continue from exactly the same point in exactly the same state, when you receive the reply. This would mean something equivalent to saving the whole call trace (the whole stack for that matter) and a lot of global variables. The ammount of complexity involved in converting ser to such a model (where such a detailed state is saved that it makes possible resuming processing at a later time) would be huge. I don't think this would be doable in finite time :-) As an alternative one could fork threads (which would save all the information involved except all the global vars.), or new processes (which would save everything). However in this case we would deal with the forking overhead. This can be attacked too (turning ser into a fork bomb). I think it's much better to start ser with lots of children processes (let's say 500, or the maximum acceptable for your machine configuration). So, I don't think async. dns would be a solution. [...] Andrei

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Re: [Serusers] problem with serctl
by Bruno Oliveira 08 Jun '05

08 Jun '05

Hi, my /var/log/syslog print: "Jun 8 09:18:58 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:18:58 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:18:58 uiara sbin/ser[2490]: Maxfwd module- initializing Jun 8 09:23:05 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:23:05 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:23:05 uiara /usr/local/sbin/ser[2534]: Maxfwd module- initializing Jun 8 09:23:27 uiara nagios: Error: Could not re-connect to database server on host '' for status data. I'll keep trying every 60 seconds... Jun 8 09:30:09 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:30:09 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:30:09 uiara /usr/local/sbin/ser[2555]: Maxfwd module- initializing" thanks for your help! On 6/8/05, Felipe Martins <fmartins(a)mundivox.com> wrote: > > Take a Look at your /var/log/syslog, and send the error message to us ....!!! > > > On Wed, 8 Jun 2005 09:27:33 -0300 > Bruno Oliveira <n1gg4s(a)gmail.com> wrote: > > > Hi, > > i have: > > ser 0.9.0 > > mysql 4.0.24 > > debian > > > > when i type command /sbin/serctl start, the serctl print this line in > > pront command: > > > > "uiara:~# /usr/local/sbin/serctl start > > > > Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" > > > > Anyone know why ? > > > > thanks for your time. > > > > -- > > []s, > > Bruno "Niggas" Oliveira > > Belo Horizonte - MG > > Msn: n1gg4s(a)gmail.com > > Icq: 176314647 > > > > "Todo o nosso descontentamento por aquilo > > que nos falta procede da nossa falta de > > gratidão por aquilo que temos." > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > Felipe Martins > Mundivox Communications > Tecnologia e Projetos > fmartins(a)mundivox.com > > Tel.: +55 +21 +3820 8839 > Cel.: +55 +21 +9823 8602 > Fax.: +55 +21 +3820 8844 > www.mundivox.com > > > -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

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[Serusers] problem with serctl
by Bruno Oliveira 08 Jun '05

08 Jun '05

Hi, i have: ser 0.9.0 mysql 4.0.24 debian when i type command /sbin/serctl start, the serctl print this line in pront command: "uiara:~# /usr/local/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" Anyone know why ? thanks for your time. -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

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[Serusers] acc.so error
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

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[Serusers] (no subject)
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

1 0

[Serusers] (no subject)
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

1 0

[Serusers] acc.so errors
by mohammed irfan 08 Jun '05

08 Jun '05

I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1)

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[Serusers] ser user in database don't exist
by Ryan Pagquil 08 Jun '05

08 Jun '05

Hi, I'm new in SER. I'm setting it up for just two days in FreeBSD in SPARC64. I run ser_mysql.sh create and everything gone fine. Then i tried to create a user using serctl add and i got an error: ERROR 1045 (28000): Access denied for user 'ser'@'localhost' (using password: YES) introducing a new user to the database failed What seems to be the problem here? FYI, i haven't yet configure ser.cfg... Thanks, Ryan

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[Serusers] Re: [Sems] do I need to upgrade to SER 0.9?
by Llanos Serna García-Conde 08 Jun '05

08 Jun '05

1 0

[Serusers] incoming call routing, voicemail on busy/no answer, load balancing, call forwarding & call display
by Chris St Denis 08 Jun '05

08 Jun '05

I am implementing a rather large scale SER server and there are some features I am having trouble figuring out how to implement. 1. Multi-number Incoming call routing. My users register with my using an id number which then needs to be mapped to 0 or more real phone numbers for incoming call routing, and 0 or 1 for outgoing call display. Is there anyway to do something like this using radius? The backend database of my system will be MS SQL server (not my choice) so radius is being used to connect to it. Multiple numbers will use distinctive rings by inserting an alert info header in with append_hf. 2. Voicemail on busy and no answer and no connect to phone. I think busy can be detected by looking for a response code but not I'm quite sure how, and I think I can use fr_timer for can't connect to phone timeout however, how do I detect no answer with a user customizable timeout? I am also not certain how to send it to the asterisk voicemail server at this point. 3. Call forwarding How can I retrieve call forwarding information from a database (preferably using radius) and forward to it? this also leads to the no answer question above. 4. Load balancing I have gotten failover to work with failure_route and t_on_failure however, I want to not just fail over but also load balance between multiple PSTN gateways. Can this be done using SRV records? round-robin DNS? is there some 'random' type command? 5. Call display I need to be able to block call display for incoming and/or outgoing calls. How can this be achieved from within SER? How can I do these things with stable SER? And, anyone know when .9 is planned to be released? Is it combing soon?

2 1

RE: [Serusers] ACK not sent
by Greg Martin 08 Jun '05

08 Jun '05

It is one call - the IP addresses are: the dialled phone (142.178.21.214), SER (142.178.218.210), the application platform (142.178.213.169) and the re-routed called party (142.178.21.182). The proxy of the application platform uses 142.178.213.201 for return messages as well just to confuse things. The cfg works fine if I don't engage the application platform in the call. > -----Original Message----- > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Tuesday, June 07, 2005 12:32 AM > To: Greg Martin; serusers(a)lists.iptel.org > Subject: Re: [Serusers] ACK not sent > > Greg, > I'm not capable of following the trace. There are several IPs > in there (142.178.21.214, 142.178.218.210, 142.178.213.169, > 142.178.21.214, 142.178.21.182). Could you download one of > the configs from http://onsip.org/ and test whether you still > have the same problems? > And if so, make sure that you only include one call with the problem. > g-) > > Greg Martin wrote: > > I'm forwarding to a server which returns an invite with changed > > completion data. I've guarded against loops by assuring the > returned > > invite isn't sent to the server again (I'm not sure that what I'm > > doing is compliant though). The correct phone rings and > when I answer > > the dialed phone thinks it is connected. The rung phone never does > > think it's connected though and a packet capture shows no ack sent > > from the dialed phone which I presume is the problem (the > phones are > > cisco 7960/7940). I'd appreciate any help. Sorry about the long > > message but felt the ngrep dump and cfg were both necessary. > > > > > ********************************************************************** > > ************ > > debug=3 > > fork=yes > > log_stderror=no > > dns=no > > rev_dns=no > > fifo="/tmp/ser_fifo" > > fifo_db_url="mysql://ser:heslo@localhost/ser" > > > > loadmodule "/usr/local/lib/ser/modules/mysql.so" > > loadmodule "/usr/local/lib/ser/modules/sl.so" > > loadmodule "/usr/local/lib/ser/modules/tm.so" > > loadmodule "/usr/local/lib/ser/modules/rr.so" > > loadmodule "/usr/local/lib/ser/modules/maxfwd.so" > > loadmodule "/usr/local/lib/ser/modules/usrloc.so" > > loadmodule "/usr/local/lib/ser/modules/registrar.so" > > > > modparam("usrloc", "db_mode", 1) > > modparam("rr", "enable_full_lr", 1) > > > > route { > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Sanity Check Section # > > > ---------------------------------------------------------------------- > > -- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too > > Many Hops"); break; }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); break; }; > > > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Record Route Section # > > > ---------------------------------------------------------------------- > > -- > > if (method != "REGISTER") { > > record_route(); > > }; > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Loose Route Section # > > > ---------------------------------------------------------------------- > > -- > > if (loose_route()) { > > route(1); > > break; > > }; > > > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Call Type Processing Section # > > > ---------------------------------------------------------------------- > > -- > > if (uri != myself) { > > route(1); > > break; > > }; > > > > > > if (method == "REGISTER") { > > route(2); > > break; > > }; > > > > > > if (method == "INVITE") { > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); break; }; > > > > if(src_ip != "142.178.213.169" > > && src_ip != "142.178.213.201" > > && src_ip != "142.178.213.170" > > && src_ip != "142.178.213.202") { > > rewritehost("142.178.213.169"); > > forward(uri:host, uri:port); > > break; > > }; > > } > > > > route(1); > > > > } > > > > route[1] { > > # > > > ---------------------------------------------------------------------- > > -- > > # Default Message Handler # > > > ---------------------------------------------------------------------- > > -- > > > > if (!t_relay()) { > > sl_reply_error(); > > }; > > > > } > > > > route[2] { > > # > > > ---------------------------------------------------------------------- > > -- > > # REGISTER Message Handler # > > > ---------------------------------------------------------------------- > > -- > > if (!save("location")) { > > sl_reply_error(); > > }; > > } > > > > > ********************************************************************** > > ***************** > > > > interface: qfe0 (142.178.218.0/255.255.254.0) > > filter: ip and ( port 5060 ) > > # > > U 142.178.21.214:50090 -> 142.178.218.210:5060 INVITE > > sip:5553331@142.178.218.210 SIP/2.0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > CSeq: 101 INVITE. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Content-Length: 249. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.169:5060 INVITE > > sip:5553331@142.178.213.169:5060;user=ip SIP/2.0. > > Max-Forwards: 10. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > CSeq: 101 INVITE. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Content-Length: 249. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > . > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 INVITE > > sip:5552221@142.178.218.210:5060 SIP/2.0. > > Via: SIP/2.0/UDP > > 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Max-Forwards: 9. > > Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 249. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 100 > trying -- > > your call is important to us. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531 > ;received=142.178.213.201. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Server: Sip EXpress router (0.9.1 (sparc64/solaris)). > > Content-Length: 0. > > Warning: 392 142.178.218.210:5060 "Noisy feedback tells: pid=14006 > > req_src_ip=142.178.213.201 req_src_port=37403 > > in_uri=sip:5552221@142.178.218.210:5060 > > out_uri=sip:5552221@142.178.21.182:5060;user=ip via_cnt==3". . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.182:5060 INVITE > > sip:5552221@142.178.21.182:5060;user=ip SIP/2.0. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKcz > K75.kDwlvStR8IbIbYqg~~531. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Max-Forwards: 8. > > Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 249. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52535 -> 142.178.218.210:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.21.182:52536 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > . > > > > # > > U 142.178.21.182:52537 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52538 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52539 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52540 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52541 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.214:50096 -> 142.178.218.210:5060 BYE > > > sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on > > SIP/2.0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:31 GMT. > > CSeq: 102 BYE. > > User-Agent: CSCO/4. > > Content-Length: 0. > > Route: > > > <sip:tc87sv01:5060;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210:5060; > > ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>, > > <sip:5552221@142.178.21.182:5060>. . > > > > # > > > > Greg Martin > > Technology & Operations > > TELUS Communication Inc. > > office: 780-493-2786 > > cell: 780-718-4139 > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > >

2 1

[Serusers] Load balancing
by Chris St Denis 08 Jun '05

08 Jun '05

I have gotten failover to work with failure_route and t_on_failure however, I want to not just fail over but also load balance. Can this be done using SRV records? round-robin DNS? is there some 'random' type command?

2 1

Re: [Serusers] add_rcv_param() for INVITEs
by Klaus Darilion 08 Jun '05

08 Jun '05

Hi Martin! As far as I understand, the problem is not the initial INVITE, but the in-dialog request, which use the fixed contact in the request URI. Correct? Do you have problems with certain clients or is it an academical question? regards, klaus Martin Koenig wrote: > Hello Klaus, > > I know that fix_nated_register works for REGISTERs, the question was > whether add_rcv_param() works for INVITEs ;) > > Answer: It does not. > > Regards, > Martin > > Klaus Darilion wrote: > >> I thin k I solved the problem using fix_nated_register instead of >> fix_nated_contact for REGISTER messages. (using nathelper) >> >> regards, >> klaus >> >> Martin Koenig wrote: >> >>> Hello, >>> >>> is it possible to use add_rcv_param() also for Contact-HF of INVITEs, >>> instead of fix_nated_contact()? >>> >>> This would make sure that the NATed End Device gets fed exactly the >>> contact as R-URI that it announced in the INVITE: >>> >>> Example R-URI in Loose-Route Requests for fix_nated_contact(): >>> >>> Ser (1.2.3.4): >>> INVITE device(a)4.5.6.7 >>> Route: 1.2.3.4;lr=on >>> >>> Device (10.0.0.1, NAT 4.5.6.7): >>> INVITE device(a)123.123.123.123 >>> >>> With add_rcv_param(): >>> >>> Ser: >>> INVITE device(a)10.0.0.1;rcv_param=4.5.6.7 >>> Route: 1.2.3.4;lr=on >>> >>> Device: >>> INVITE device(a)10.0.0.1 >>> >>> The problem is that the End-Device, in the fix_nated_contact() >>> callflow, has to accept a Request that is not really for this device >>> (IP in Request-URI != IP of the Device). Imagine a Ser checkip uri == >>> myself. This would fail without knowing the public IP of the NAT and >>> hard-conding it into Ser.cfg. >>> >>> Does Ser support the rcv_param in Request-URI also? >>> >>> What do you think? Any experience on this? >>> >>> Regards, >>> Martin >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> > >

1 0

[Serusers] ERROR: t_get_trans_ident: no transaction found
by Roger Lewau 08 Jun '05

08 Jun '05

Whenever I make a call I get these entries in the log.... Jun 7 17:56:29 ns2 ser[520]: WARNING: script writer didn't release transaction Jun 7 17:56:42 ns2 ser[522]: ERROR: t_get_trans_ident: no transaction found Jun 7 17:56:42 ns2 ser[522]: assemble_msg: t_get_trans_ident failed Jun 7 17:56:42 ns2 ser[522]: ERROR:tm:t_write_req: Error int assemble_msg Jun 7 17:56:42 ns2 ser[522]: ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established If the user is online I get connected properly, if user is offline I get connected to sems. So It seems to be working after all. But I had this config running for three days then I got a lot of errors regarding t_newtran and out of memory... and phones lost their registration. Also does any have any idea where to put t_release() to avoid the first WARNING message... This is my ser.cfg debug=3 fork=yes log_stderror=no listen=212.247.91.237 port=5060 children=4 alias=sip.serverhallen.com dns=no rev_dns=no fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" #LADDA IN DE MODULER VI BEHÖVER #--------------------------------------------------------------------------- ------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" #MySQL database support loadmodule "/usr/local/lib/ser/modules/sl.so" #Stateless replies loadmodule "/usr/local/lib/ser/modules/tm.so" #Transaction Management loadmodule "/usr/local/lib/ser/modules/rr.so" #Routing and Record-Routing loadmodule "/usr/local/lib/ser/modules/maxfwd.so" #Max-Forwards check loadmodule "/usr/local/lib/ser/modules/usrloc.so" #User location support loadmodule "/usr/local/lib/ser/modules/registrar.so" #Registrar loadmodule "/usr/local/lib/ser/modules/auth.so" #Common functions for digest authentication loadmodule "/usr/local/lib/ser/modules/auth_db.so" #Database support for digest authentication loadmodule "/usr/local/lib/ser/modules/uri.so" #Various URI checks loadmodule "/usr/local/lib/ser/modules/uri_db.so" #URI checking using DB loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #RTP Mediaproxy for NATed UAs loadmodule "/usr/local/lib/ser/modules/nathelper.so" #Functions to handle NAT traversal loadmodule "/usr/local/lib/ser/modules/textops.so" #Message Textual Operations loadmodule "/usr/local/lib/ser/modules/avpops.so" #avp operations, pseudo-variable support loadmodule "/usr/local/lib/ser/modules/domain.so" #checks if a host part of an URI is "local" or not. loadmodule "/usr/local/lib/ser/modules/permissions.so" #determine if a call has appropriate permission to be established #VARIABLER FÖR DATABAS OCH HANTERINGA AUTENTISIERING #--------------------------------------------------------------------------- ------------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) #VARIABLER FÖR HANTERING AV NATade UAs #--------------------------------------------------------------------------- -------------------- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) #VARIABLER FÖR HANTERING AV PSTN #--------------------------------------------------------------------------- -------------------- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #MEDDELANDEHANTERING AV SIP MEDDELANDEN #--------------------------------------------------------------------------- -------------------- #Alla meddelanden kommer behandlas enligt detta regelverk i sekventiell ordning route { # ----------------------------------------------------------------- # Sanity Check Section, Kontrollera hur många gånger ett meddelande tagits emot. # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section, Sektion som lägger till vår egen sip server i routing kedjan så att svaren kommer tillbaka till oss. # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.247.91.237:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section, Denna sektion hanterar nedtagning av samtal # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { # end_media_session(); # if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; # ----------------------------------------------------------------- # Loose Route Section, # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section, om meddelandet inte skulle skickas vidare är det förmodligen riktat till SER # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { # switch to statefull mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; log("NO MATCH"); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler, Standardhantering av Meddelanden # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()){ if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:[+|00][0-9]*@") { route(4); break; }; if (uri=~"^sip:0[1-9]*@") { route(4); break; }; if (uri=~"sip:100.*@") { if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems"); }; break; }; if (!lookup("location")) { #Send caller to users answering machine if uri is local if(!t_write_req("/tmp/am_fifo","voicemail")) { t_reply("500","error contacting sems"); }; break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost(" 192.168.0.1 "); avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler, aktiverar proxyn # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section, hanterar acks # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

RE: [Serusers] Mediaproxy in LAN (?)
by a cairo 08 Jun '05

08 Jun '05

Thanks, but if all clients connect to ser how balancing happen from mediaproxy? -----Original Message----- From: "Shaikat Mahmud" <shaikat(a)stitel.com> To: "'a cairo'" <a.cairo(a)scsitalia.it>, "'Giudice, Salvatore'" <Salvatore.Giudice(a)FMR.COM>, "'bogdan aux'" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> Date: Sat, 4 Jun 2005 16:29:36 +0600 Subject: RE: [Serusers] Mediaproxy in LAN (?) > Hi, > > In mediaproxy, README file has all the architectural information. > > Is mediaproxy between client and ser? > No, its some thing like below, > > > Ua1--------------> SER-------------->UA2 > | > | (Mediaproxy module requesting for a session) > | > | > | > MediaProxy > > > Then RTP flow will be like following: > Ua1 <-------> MediaProxy <-------> Ua2 > > Read the README file for details. > shaikat > > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of a cairo > Sent: Saturday, June 04, 2005 4:14 PM > To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org > Subject: RE: [Serusers] Mediaproxy in LAN (?) > > Hi all, > i don't understand architecture of mediaproxy with ser ! > Is mediaproxy between client and ser? > Thanks > > -----Original Message----- > From: "Giudice, Salvatore" <Salvatore.Giudice(a)FMR.COM> > To: "bogdan aux" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> > Date: Tue, 17 May 2005 20:10:46 -0400 > Subject: RE: [Serusers] Mediaproxy in LAN (?) > > > I am trying to route a call from a non-routable 10.x.x.x network to a > > phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN > to > > SIP from my asterisk to an x-lite > > > > I found some references indicating that I should call: > > > > > > if ( src_ip = IP_GATEWAY) > > { > > force_rport(); > > fix_contact(); > > use_media_proxy(); > > }; > > > > Supposedly, you need the force_rport and the fix_contact to put media > > proxy in the middle of the two rtp sessions, but I have et to see > this > > actually work. Has anyone been able to pass a call through media > proxy > > from a GW on a non-routable to a routable network, or even between > two > > ip's on the same network as the mediaproxy? > > > > If so, please share your configs. > > > > -----Original Message----- > > From: bogdan aux [mailto:aux1d@yahoo.com] > > Sent: Friday, May 13, 2005 6:16 AM > > To: serusers(a)lists.iptel.org > > Subject: [Serusers] Mediaproxy in LAN (?) > > > > I want to route RTP packets between 2 computers in the > > same network through Mediaproxy. I know this may have > > no sense, but it's just a test configuration. > > The call use_media_proxy() seems to have no effect, > > the RTP packets go directly from one computer to > > another. > > My guess is mediaproxy makes a test on the IPs (they > > are 192.168.147.2 and 192.168.147.3) and refuses to > > proxy the call. > > Is mediaproxy working just between different networks > > or it's an error in my configuration file? > > > > > > > > > > Yahoo! Mail > > Stay connected, organized, and protected. Take the tour: > > http://tour.mail.yahoo.com/mailtour.html > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

3 2

[Serusers] Re-routing to same device in failure_route
by Karun Chemudugunta 07 Jun '05

07 Jun '05

Hi Group, Cisco 5300 -----> SER ----> Xten softphone-A ----> Failed call with error code >400 -------> try to route to Xten softphone A(same as previous) I am trying to route calls to softphones in such way that, if call failed by some reason (>400), trying to route second time to the same soft phone using append_brach() and t_relay() like below. But, soft phone is unable process INVITE message for some reason. But, if I forward failed call different soft phone, call was successful. Am I doing any thing wrong in config ??? Thanks for your help in advance. Regards, Karun ########################################################################################## # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=9 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #alias=voip.cafe.bevocal.com #alias=66.77.14.238 alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/lib/ser/modules/auth.so" #loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol # record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { log(1,"Inside failure_route five"); revert_uri(); append_branch(); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); }; } ######################################################################################

2 1

[Serusers] Re: [Serdev] ENUM issues - req. for discussion
by Samuel Osorio Calvo 07 Jun '05

07 Jun '05

Hi Klaus, responses inline...(command and file names are taken from my memory, so don't pay much attention at the spelling) Samuel. Unclassified. >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 06/07/05 01:43PM >>> >Hi all! > >Some time of using ser with ENUM revealed several problems which I would >like to dicuss with you. Be aware - this email is long! > >ENUM is a wonderful thing for call routing nevertheless, as it is DNS >based there are some important things an ENUM aware application has to >consider: > 1. You never know how long the lookup takes > 2. You never know if the lookup will fail or succeed > 3. If the lookup was successful, you never may trust the result> > >I will explain this points in detail know: > >1+2: >Using enum, the application is giving control to the DNS resolver of the >OS and the DNS infrastructure. Thus, the ser thread which performs the >ENUM lookup will be blocked until there is a result from the system's >DNS resolver. > >If DNS is slow, or misconfigured (e.g. a zone is delegated to a >nameserver which is down), the thread will be blocked for several >seconds. E.g. if you use debian woody and 2 nameservers in >/etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS >allows configuration of the DNS timeouts. Nevertheless, you have to >consider that a ser thread will be blocked up to 20 seconds. This has >impacts on your configuration: I don't know the details but would it be really difficult to use an asyncrhonous resolver, such as resiprocate SIP stack does with ARES?? Besides exec_* calls, the main SER's performance bottelneck is the DNS resolving step thus it would be a great improvement adding asyncronous DNS queries. >Typically, you use some kind of the following logic: >if (uri =~ "+[0-9].") { > if (enum_lookup()) { > t_relay(); break; > } else { > forward to PSTN gateway; break; > } >} > >Thus, the INVITE will be received and the ENUM lookup will be performed. >If the lookup will take longer than 0.5s, the SIP client will start >restransmitting the INVITE. Thus, another thread will process this >INVITE and enother ENUM lookup will be performed. After several seconds, >all of ser's threads will be blocked with ENUM lookups and your SIP >proxy will not handle any requests until the DNS query times out. Thus, >it is very easy to generate a DoS attack against the proxy. >Another funny thing is, that the SIP client will detect a proxy error >and hangs up, but the INVITEs are still processed in the SIP proxy and >after the timeout forwarded to the PSTN gateway. > >A solution to stop the retransmission is to immediatle >sl_send_reply("100","Trying"), But this rises another problem. Now, if >the caller hangs up before the DNS timeout, the SIP client will send >CANCEL (as it received 100) to the SIP proxy. But the SIP proxy can not >cancel the transaction, as it is not genereated yet - the INVITE thread >is still waiting for the ENUM lookup and the transaction will be >generated after the ENUM lookup (after 20s timeout). Thus, we still end >up with an INVITE forwarded to the PSTN gateway although the SIP client >already hang up. > >I thought of using t_newtran and t_forward_nonack_uri instead of t_realy >to generate the transaction before doing the ENUM lookup. Thus, the >thread which will process the CANCEL should find a transaction and stop >it. But will this really prevent the INVITE sent to the PSTN gateway, >once the DNS times out? (not tested) I think the best way would be to add a reply(100) AND t_forward_nonack, something like: sl_send_reply("100","Trying"); t_newtran(); uri =~ "+[0-9].") { if (enum_lookup()) { t_forward_nonack(); break;} else{ t_forward_nonac();break; } But it adds a complexity level which will lead to many non-working config files......but I vote for it! > >btw: this problem does not only occour for ENUM lookups, but for all DNS >lookups (SRV, A, AAAA). > >3: >If the ENUM lookup succeeds, you never may trust the result. It may be a >invalid SIP URI, or a tel: URI, or anything else a user puts into its >NAPTRs. This may result in a failed transaction, or like revealed at the >ENUM plugtest in failed accounting. Even worse, maybe it is possible to >complete crash ser using realy bad formated URIs? > >Thus you can't avoid doing some URI checks against the URI received from >the ENUM lookup. Perfomance issues are no valid arguements! Once I give >control to external services (DNS, radius, exec), the perfomance >penalties due to parsing the SIP URI are much more less than due to the >ENUM lookup. > >In case of ser, I would do the URI parsing in the ENUM module, or maybe >generate a dedicated function/module for checking SIP URIs inside the >routing logic. Thus, I can also check the result of exec calls. Since URI check is a really global action, it should be implemented in SER's core, providing commands to the config file (something like uri_check(To|From|Req-URI|Contact) ) and functions to be called from any module. > >I'm eagerly waiting for your opinions. > >regards, >Klaus _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

3 2

[Serusers] Re: RADIUS failure for long domains returned by ENUM queries
by Adrian Georgescu 07 Jun '05

07 Jun '05

Juha, what if the response from DNS is greater then 253 bytes? Juha Heinanen wrote: >adrian, > >the value of the attribute that you try to add is 202 bytes long. >my /usr/include/radiusclient.h defines > >#define AUTH_STRING_LEN 128 /* maximum of 253 */ > >typedef struct value_pair >{ > char name[NAME_LENGTH + 1]; > int attribute; > int type; > UINT4 lvalue; > char strvalue[AUTH_STRING_LEN + 1]; > struct value_pair *next; >} VALUE_PAIR; > >thus the value is too long and would fail. > >on the other hand, my /usr/include/radiusclient-ng defines: > >#define AUTH_STRING_LEN 253 /* maximum of 253 */ > >which one are you using in your test setup? > >-- juha > >

3 2

[Serusers] ENUM issues - req. for discussion
by Klaus Darilion 07 Jun '05

07 Jun '05

Hi all! Some time of using ser with ENUM revealed several problems which I would like to dicuss with you. Be aware - this email is long! ENUM is a wonderful thing for call routing nevertheless, as it is DNS based there are some important things an ENUM aware application has to consider: 1. You never know how long the lookup takes 2. You never know if the lookup will fail or succeed 3. If the lookup was successful, you never may trust the result I will explain this points in detail know: 1+2: Using enum, the application is giving control to the DNS resolver of the OS and the DNS infrastructure. Thus, the ser thread which performs the ENUM lookup will be blocked until there is a result from the system's DNS resolver. If DNS is slow, or misconfigured (e.g. a zone is delegated to a nameserver which is down), the thread will be blocked for several seconds. E.g. if you use debian woody and 2 nameservers in /etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS allows configuration of the DNS timeouts. Nevertheless, you have to consider that a ser thread will be blocked up to 20 seconds. This has impacts on your configuration: Typically, you use some kind of the following logic: if (uri =~ "+[0-9].") { if (enum_lookup()) { t_relay(); break; } else { forward to PSTN gateway; break; } } Thus, the INVITE will be received and the ENUM lookup will be performed. If the lookup will take longer than 0.5s, the SIP client will start restransmitting the INVITE. Thus, another thread will process this INVITE and enother ENUM lookup will be performed. After several seconds, all of ser's threads will be blocked with ENUM lookups and your SIP proxy will not handle any requests until the DNS query times out. Thus, it is very easy to generate a DoS attack against the proxy. Another funny thing is, that the SIP client will detect a proxy error and hangs up, but the INVITEs are still processed in the SIP proxy and after the timeout forwarded to the PSTN gateway. A solution to stop the retransmission is to immediatle sl_send_reply("100","Trying"), But this rises another problem. Now, if the caller hangs up before the DNS timeout, the SIP client will send CANCEL (as it received 100) to the SIP proxy. But the SIP proxy can not cancel the transaction, as it is not genereated yet - the INVITE thread is still waiting for the ENUM lookup and the transaction will be generated after the ENUM lookup (after 20s timeout). Thus, we still end up with an INVITE forwarded to the PSTN gateway although the SIP client already hang up. I thought of using t_newtran and t_forward_nonack_uri instead of t_realy to generate the transaction before doing the ENUM lookup. Thus, the thread which will process the CANCEL should find a transaction and stop it. But will this really prevent the INVITE sent to the PSTN gateway, once the DNS times out? (not tested) btw: this problem does not only occour for ENUM lookups, but for all DNS lookups (SRV, A, AAAA). 3: If the ENUM lookup succeeds, you never may trust the result. It may be a invalid SIP URI, or a tel: URI, or anything else a user puts into its NAPTRs. This may result in a failed transaction, or like revealed at the ENUM plugtest in failed accounting. Even worse, maybe it is possible to complete crash ser using realy bad formated URIs? Thus you can't avoid doing some URI checks against the URI received from the ENUM lookup. Perfomance issues are no valid arguements! Once I give control to external services (DNS, radius, exec), the perfomance penalties due to parsing the SIP URI are much more less than due to the ENUM lookup. In case of ser, I would do the URI parsing in the ENUM module, or maybe generate a dedicated function/module for checking SIP URIs inside the routing logic. Thus, I can also check the result of exec calls. I'm eagerly waiting for your opinions. regards, Klaus

5 14

[Serusers] outbound proxy
by harry gaillac 07 Jun '05

07 Jun '05

Hello ser users, I wish all sip message are handle by ser. phone1 -------------------------- | ser+mediaproxy|------internet phone2 -------------------------- In order to send all sip messages via ser I set outbound sip proxy (ser/public ip) for sip clients. However ser don't route invite message for other domains.Why? Harry # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.9.14 # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) # --mediaproxy param -- modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # --domain params -- modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") { record_route(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("3")) { setflag(6); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # Requests to others domains if (!is_from_local()) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_from_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { end_media_session(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (nat_uac_test("3")) { setflag(6); force_rport(); fix_contact(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("3")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_contact(); use_media_proxy(); }; if (!is_from_local()) { route(1); break; }; # SIP destinations from PSTN if (src_ip==192.168.0.50 && allow_trusted()) { route(4); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; }; # SIP destinations to PSTN if (uri=~"^sip:1013@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$") { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); break; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] Re: Welcome to the "Serusers" mailing list (Digest mode)
by gsalvucci＠ing.unrc.edu.ar 07 Jun '05

07 Jun '05

Mensaje citado por serusers-request(a)lists.iptel.org: > Welcome to the serusers(a)lists.iptel.org mailing list! > > To post to this list, send your email to: > > serusers(a)lists.iptel.org > > General information about the mailing list is at: > > http://lists.iptel.org/mailman/listinfo/serusers > > If you ever want to unsubscribe or change your options (eg, switch to > or from digest mode, change your password, etc.), visit your > subscription page at: > > http://lists.iptel.org/mailman/options/serusers/gsalvucci%40ing.unrc.edu.ar > > > You can also make such adjustments via email by sending a message to: > > serusers-request(a)lists.iptel.org > > with the word `help' in the subject or body (don't include the > quotes), and you will get back a message with instructions. > > You must know your password to change your options (including changing > the password, itself) or to unsubscribe. It is: > > hipolita > > Normally, Mailman will remind you of your iptel.org mailing list > passwords once every month, although you can disable this if you > prefer. This reminder will also include instructions on how to > unsubscribe or change your account options. There is also a button on > your options page that will email your current password to you. > > ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

1 0

[Serusers] setting up SER
by Tomasz Gingold 07 Jun '05

07 Jun '05

Hello, i´m fairly new in SER. After setting up the SER i tried to register at SER with the kphone. Inside the kphone i set up the Outbound proxy with the adress of the SER. When i try to REGISTER with the kphone at SER i get always also a REGISTER message back from the SER. Kphone SER -----REGISTER---------> <----REGISTER---------- (the first REGISTER follow 3xREGISTER back) <----REGISTER---------- <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- ... <----403 Timeout------- It looks like the SER doesn't go into the routing logic. I modify the ser.cfg in this way to check this. route { sl_send_reply("420","Bad Extension"); After this nothing has changed in the packet flow described before. No 420 replys are beeing send back. I know that this is not a detailed description of my problem.Anyhow i hope that anybody can give me some tips. thanks in advance tomasz g. -- Geschenkt: 3 Monate GMX ProMail gratis + 3 Ausgaben stern gratis ++ Jetzt anmelden & testen ++ http://www.gmx.net/de/go/promail ++

3 3

[Serusers] Testing with SIP and ENUM
by Adrian Georgescu 07 Jun '05

07 Jun '05

For those of you who need accounts to test their own SER proxies from outside their domains I have setup this easy registration service http://sip2sip.info You get a SIP account under sip2sip plus a UPT +87810 ENUM number and you may call outside local domain with it. There are some mediaproxies that do the NAT traversal, video streams included. And BTW check the SNOM software phone from http://snom.com/download/snom360-3.60j.exe it is by far the easiest SIP software I had configured on my desktop. Regards, Adrian

1 0

[Serusers] help with: can't read reply from a RTP proxy
by jeff kwong 07 Jun '05

07 Jun '05

Hi everybody! Im a newbie with ser and I have just joined the list. You see I want to run a ser+nathelper+rtpproxy system in my linux box. My primary task is to build a sip outbound proxy. As a start I used the nathelper.cfg file in sip_router from the cvs repository. I also downloaded and installed the latest ser and rtpproxy from cvs. I always run rtpproxy first and I can verify its is running with ps. However when I execute ser with nathelper.cfg I keep getting the following message: 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) WARNING: using only the first listen address (no fork) 2(0) INFO: fifo process starting: 15323 1(15322) ERROR: send_rtpp_command: can't read reply from a RTP proxy 1(15322) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 1(15322) WARNING: rtpp_test: can't get version of the RTP proxy 1(15322) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily 2(15323) ERROR: send_rtpp_command: can't read reply from a RTP proxy 2(15323) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 2(15323) WARNING: rtpp_test: can't get version of the RTP proxy 2(15323) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily 2(15323) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(15323) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(15321) ERROR: send_rtpp_command: can't read reply from a RTP proxy 0(15321) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 0(15321) WARNING: rtpp_test: can't get version of the RTP proxy 0(15321) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily and thus I cannot make rtp calls. I know similar problems have been posted before and the most frequent suggestion was to get the latest versions of ser and rtpproxy like the one below. 1) Get ser from CVS, build, install cvs -d:pserver:anonymous at cvs.berlios.de <http://lists.iptel.org/mailman/listinfo/serusers>:/cvsroot/ser co sip_router 2) Get rtpproxy from CVS, build, install cvs -d:pserver:anonymous at cvs.berlios.de <http://lists.iptel.org/mailman/listinfo/serusers>:/cvsroot/ser co rtpproxy 3) Edit ser.cfg to add modparam("nathelper","rtpproxy_sock", "/var/run/rtpproxy.sock") I did as instucted but still the same results. What could I be missing? To those who have successfully configured a similar setup, can I ask what version of ser and rtpproxy are compatible? Thanks in advance! jeff

2 1

[Serusers] Configuration example of lcr.so
by Felipe Martins 07 Jun '05

07 Jun '05

Hi everyone, Has anybody an example of ser.cfg using lcr.so module doing routing logics querying a MySQL DB ? If so, can anyone send it to me ? I have read all the documentation but I'm still confusing about it... Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

1 0

[Serusers] 4/8 port GW
by Iqbal 07 Jun '05

07 Jun '05

Hi Does anyone have any good experience with 4/8 port gateways and connecting them to SER. If so which are recommended, they must use user/pass to auth I guess. Iqbal

3 2

[Serusers] domain/realm routing
by raid＠gmx.net 07 Jun '05

07 Jun '05

hi all example: uac1 with number 12345 and realm my1.mydomain.com uac2 with number 54321 and realm my2.mydomain.com and so on i use enum to write the called brunch to the realm this funktion is ok (if someone call 12345/123, enum rewrite the uri to 123(a)my1.mydomain.com) but ser says no user 123(a)my1.mydomain.com if i change in enum zonefile my1.mydomain.com to the ip address of the uac ser send the call to the uac1, but there is no connect because the uac1 is behind nat the next problem is if the user has a dynamic ip is there a way to bind an uac/user to an realm (all calls 12345/branches route to uac1(a)my1.mydomain.com) i hope you understand my problem sorry for my bad english :( regards raid -- Weitersagen: GMX DSL-Flatrates mit Tempo-Garantie! Ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl

3 4

[Serusers] Loose routing question
by Michael Ulitskiy 07 Jun '05

07 Jun '05

Hello, I'm trying to comprehend loose routing concept and I have a question that concerns me. As far as I understand loose routing says that if there're Route headers in a message it should be forwarded according to the URIs set in Route headers. I thought that this is true only within a dialog, but RFC3261 (part 16.6) says: "Requests establishing a dialog may contain a preloaded Route header field." Also SER manual says: " the failure not to include loose routing in your scripts may lead to infinite loops. Make sure that you include the following script fragment immediately after request sanity checks" and provide the following piece of code: if (loose_route()) { t_relay(); break; }; which as far as I understand unconditionally forwards message if Route header is present. So I'm wondering what about security? If I follow this guidelines how I would shield my PSTN gateway if anyone can construct message and pre-load it with URI of my gateway and all my proxies must honor it. For example I have a PSTN gateway on ip address 10.1.1.5 and proxy on 10.1.1.10 that supposed to interface outside world. So I guess if someone construct a message like this: INVITE sip:12345@somewhere.com SIP/2.0 ... Route: <sip:12345@10.1.1.5;lr> my proxy will forward it to PSTN gateway and it will make outbound call. Is this true? Please enlighten me on this. Thank you, Michael

7 9

[Serusers] do I need to upgrade to SER 0.9?
by Llanos Serna García-Conde 07 Jun '05

07 Jun '05

1 0

[Serusers] MAX_LINE_SIZE
by Llanos Serna García-Conde 07 Jun '05

07 Jun '05

1 0

Re: [Serusers] Can I use realm name without DNS SRV
by Greger V. Teigre 07 Jun '05

07 Jun '05

Sorry, Antonio. This is getting too much of a special case that I can devote any time to it. I would suggest you clean up and concentrate on a pure IP setup (also in the user clients), which, BTW, I have done at several occations, even without an Internet connection. Remember that SER does a reverse lookup (gethostbyname) to resolve it's own IP address. g-) Antonio Cano damas wrote: > Dear Greger, > > I went to fast into this conclusion :( > > I've made rewritehostport("Public_IP:5070") because if I use the > private_ip SER does nothing :? > > The problem now is in Asterisk or Maybe in the network config, I think > is second. > > In my network scenario the Redstar servir is connected to the internet > throught an ADSL Router, with all ports redirected to it: > Internet ----> Public_IP (Router ADSL) Private_IP <---- (Redstar) > > With Tethereal We can see this log: > 11.055288 192.168.1.5 -> Public_IP SIP/SDP Request: INVITE > sip:213@Public_IP:5070, with session description > 11.055982 192.168.1.1 -> 192.168.1.5 ICMP Destination > unreachable (Port unreachable) > > The problem is that packet try to go throw the Internet and the Router > doesn't know how to give it. Due to the Proxy and Asterisk are both > into the same machine I doesn't need that travel :) > > In /etc/hosts I've got: > 192.168.1.5 Public_IP redstar.organization.org redstar > 192.168.1.5redstar.organization.org > > For iptables I've added: > iptables -t nat -A PREROUTING -d Public_IP -p udp --dport 5060:5070 > -j DNAT --to 192.168.1.5 > > After the changes of /etc/hosts I rebooted the server. Now I make some > tests with traceroute: > > redstar:~# traceroute redstar > traceroute to Public_IP (192.168.1.5), 30 hops max, 38 byte packets > 1 80 (192.168.1.5) 0.268 ms 0.058 ms 0.750 ms > > redstar:~# traceroute Public_IP > traceroute to Public_IP (Public_IP), 30 hops max, 38 byte packets > 1 192.168.1.1 (192.168.1.1) 0.634 ms 0.451 ms 0.361 ms > > > Do you have some idea about that? > > Kind regards, > Antonio F. Cano > > > > Greger V. Teigre wrote: > >> ;-) >> Antonio Cano damas wrote: >> >>> wowwwww, a lot of thanks it goes :D >>> >>> >>> Greger V. Teigre wrote: >>> >>>> SER will try to resolve using DNS SRV and A lookups unless you >>>> specify IP addresses, so: >>>> route[2] >>>> { >>>> #Host and port where Asterisk is listening, sip.conf >>>> rewritehostport("192.168.1.5:5070"); >>>> route(1); >>>> break; >>>> } >>>> g-) >>>> >>>> Antonio Cano damas wrote: >>>> >>>>> Hi, >>>>> >>>>> That seems to start working :). I've made the next changes: >>>>> listen="redstar.organization.org" >>>>> alias="192.168.1.5" >>>>> alias="organization.org" >>>>> alias="redstar.organization.org" >>>>> >>>>> check_via=no # (cmd. line: -v) >>>>> dns=no # (cmd. line: -r) >>>>> rev_dns=no # (cmd. line: -R) >>>>> >>>>> And the proper updates into MySQL Database for domain like >>>>> 'redstar.organization.org'. >>>>> >>>>> Now I've got happening something strange, my UAs x-lite and >>>>> grandstream can register but Asterisk doesn't (at first look). >>>>> Looking the URIS I've seen that X-lite generate >>>>> @192.168.1.5redstar.organization.org while Asterisk generate >>>>> @redstar.organization.org. Finally I've got registering the >>>>> Asterisk using fromdomain= 192.168.1.5restar.organization.org :-/ >>>>> >>>>> My next step is try to make a call from X-Lite to one extension of >>>>> Asterisk. In the SER.cfg: >>>>> A) Into the INVITE process section: >>>>> if (uri =~ "^sip:(2[0-9][0-9])@*" ) { >>>>> log(1,"Al asterisk\n"); >>>>> route(2); >>>>> break; >>>>> }; >>>>> B) >>>>> route[2] >>>>> { >>>>> #Host and port where Asterisk is listening, sip.conf >>>>> rewritehostport("192.168.1.5redstar.organization.org:5070"); >>>>> route(1); >>>>> break; >>>>> } >>>>> >>>>> route[1] >>>>> { >>>>> # if client or server know to be behind a NAT, enable relay >>>>> if (isflagset(6)) { >>>>> #log(1, "Pasando por force_rtp_proxy\n"); >>>>> append_hf("P-Hint: Pasando por RTP_PROXY\r\n"); >>>>> force_rtp_proxy(); >>>>> }; >>>>> >>>>> # labeled all transaction for accounting >>>>> #setflag(4); >>>>> >>>>> # send it out now; use stateful forwarding as it works >>>>> reliably # even for UDP2TCP >>>>> if (!t_relay()) { >>>>> sl_reply_error(); >>>>> }; >>>>> } >>>>> >>>>> With that the INVITE petition gets an Request Timeout response, >>>>> making a little of ngrep: >>>>> >>>>> U IP_Client_UA:5060 -> 192.168.1.5:5060 >>>>> INVITE sip:213@192.168.1.5redstar.organization.org SIP/2.0..Via: >>>>> SIP/2.0/UDP >>>>> 192.168.0.6:5060;rport;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: >>>>> >>>>> >>>>> Anto >>>>> nio F. Cano >>>>> <sip:10101@redstar.organization.org>;tag=1817289323..To: >>>>> <sip:213@redstar.organization.org>..Contact: >>>>> <sip:10101@192.168.0.6:5060>..Call-ID: 4F13C7CB-D >>>>> 6AD-11D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 >>>>> INVITE..Max-Forwards: 70..Content-Type: >>>>> application/sdp..User-Agent: X-Lite release 1103m..Content-Len >>>>> gth: 263....v=0..o=10101 5457996 5458244 IN IP4 >>>>> 192.168.0.6..s=X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 >>>>> RTP/AVP 3 0 8 98 101..a=rtpmap:0 pcmu/80 00..a=rtpmap:8 >>>>> pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 >>>>> iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >>>>> 0-15.. >>>>> # >>>>> >>>>> U 192.168.1.5:5060 -> IP_DNS_SERVER:5070 >>>>> INVITE sip:213@192.168.1.5redstar.organization.org:5070 >>>>> SIP/2.0..Record-Route: >>>>> <sip:213@192.168.1.5;ftag=1817289323;lr=on>..Via: SIP/2.0/UDP >>>>> 192.168.1.5;branc h=z9hG4bK151f.afc77ca4.0..Via: SIP/2.0/UDP >>>>> 192.168.0.6:5060;received=IP_Client_UA;rport=5060;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: >>>>> >>>>> >>>>> Antonio >>>>> F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: >>>>> <sip:213@redstar.organization.org>..Contact: >>>>> <sip:10101@IP_Client_UA:5060>..Call-ID: 4F13C7CB-D6AD-1 >>>>> 1D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 >>>>> INVITE..Max-Forwards: 69..Content-Type: application/sdp..User-Agent: >>>>> X-Lite release >>>>> 1103m..Content-Length: >>>>> 283..P-Hint: Nat uac_test=3 ..P-Hint: Metodo INVITE corrigiendo >>>>> SDP..P-Hint: Pasando por RTP_PROXY....v=0..o=10101 5457996 5458244 >>>>> IN IP4 192.168.0.6..s= >>>>> X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 RTP/AVP 3 0 8 >>>>> 98 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 >>>>> gsm/8000..a=rtpmap:98 iLBC >>>>> /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >>>>> 0-15..a=direction:active.. >>>>> >>>>> Why SER sends the message to IP_DNS_SERVER in place of >>>>> 192.168.1.5redstar.organization.org? >>>>> >>>>> What's happen here? I don't understand nothing. What I make wrong? >>>>> The @192.168.1.5redstar.organization.org doesn't seem to be so >>>>> elegant is there any way to make it like redstar.organization.org >>>>> >>>>> Kind regards and thanks for your time, >>>>> Antonio F. Cano >>>>> >>>>> Greger V. Teigre wrote: >>>>> >>>>>> You can turn off SRV in X-Lite and Grandstream (Use DNS SRV >>>>>> option) And use SIP proxy: fully qualified domain name (same as >>>>>> outbound) g-) >>>>>> Antonio Cano damas wrote: >>>>>> >>>>>>> Hello all, >>>>>>> >>>>>>> First of all excuses for my bad english. I was looking into the >>>>>>> list and doesn't find nothing similar to my problem maybe I >>>>>>> didn't use the correct keywords. >>>>>>> >>>>>>> I've got a Proxy that is behaind and Public IP and I've got a >>>>>>> subdomain name pointing to this machine. But the DNS server >>>>>>> isn't mine and I can't add a SRV record, I've redirected all >>>>>>> petitions to this server and in that way if I make a petition >>>>>>> of anykind service (ssh, web, ...) to the subdomain.name.com it >>>>>>> comes into the machine. The problem is that all de UA i try >>>>>>> (X-Lite, Grandstream and Asterisk) makes DNS SRV queries and i >>>>>>> don't know how to avoid this. For example: >>>>>>> >>>>>>> My subdomain is redstar.organization.org and I'd like to use the >>>>>>> realm organization.org >>>>>>> >>>>>>> + File /etc/hosts >>>>>>> 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org >>>>>>> redstar >>>>>>> >>>>>>> + In /etc/init.d/ser I've got this params line: >>>>>>> PARAMS="-P $PIDFILE -u root -g root -l >>>>>>> redstar.organization.org" --->I try using -l >>>>>>> organization.org, but the server said to me: >>>>>>> 0(0) ERROR: udp_init: bind(5, 0x80c8cc0, 16) on >>>>>>> IP_Organization.org: Cannot assign requested address >>>>>>> >>>>>>> + In /etc/ser/ser.cfg added this: >>>>>>> alias="organization.org" >>>>>>> alias="redstar.organization.org" >>>>>>> >>>>>>> + MySQL tables the domain field value is 'organization.org' >>>>>>> >>>>>>> When I start the SER it tells me: >>>>>>> Restarting ser: serListening on >>>>>>> Public_IP_Of_Redstar [192.168.1.5]:5060 >>>>>>> Aliases: redstar:5060 redstar.organization.org:* >>>>>>> organization.org:* The UA Grandstream config is: >>>>>>> SIP Server: organization.org >>>>>>> Outbound Proxy: redstar.organization.org >>>>>>> >>>>>>> And the received answer is a 478 Unresolveable destination. Can >>>>>>> anyone help me? What can I do? I'm a little desesperated :( >>>>>>> >>>>>>> Thanks in advance, >>>>>>> Antonio F. Cano >>>>>>> >>>>>>> Ngrep log result into redstar.organization.org machine: >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> # >>>>>>> >>>>>>> Why this 'sip:192.168.1.5organization.org'? >>>>>>> >>>>>>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!.. >>>>>>> >>>>>>> >>>>>>> ###################################################################### >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> ######################################################### >>>>>>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> om.B..@..........:..... >>>>>>> >>>>>>> # >>>>>>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!.. >>>>>>> >>>>>>> ########## >>>>>>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> om.B..@..........:..... >>>>>>> >>>>>>> ################################################################################################################## >>>>>>> >>>>>>> >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> ########################################################################################################################### >>>>>>> >>>>>>> >>>>>>> >>>>>>> U 192.168.1.5:5060 -> UA_Public_IP:62303 >>>>>>> SIP/2.0 478 Unresolveable destination (478/TM)..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083 >>>>>>> fbe2c60e48e32;rport=62303;received=IP_Client_UA..From: "Antonio >>>>>>> F. Cano (iGT)" <sip:10101@organization.org; >>>>>>> user=phone>;tag=b511698681c5146b..To: >>>>>>> <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 >>>>>>> f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: >>>>>>> 100 REGISTER..Server: Sip EXpress route >>>>>>> r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 >>>>>>> 80.38.244.122:5060 "Noisy feedback tells: >>>>>>> pid=8023 req_src_ip=IP_Client_UA req_src_port=62303 >>>>>>> in_uri=sip:192.168.1.5organization.org out_uri=sip:192. >>>>>>> 168.1.5organization.org >>>>>>> via_cnt==1".... >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Serusers mailing list >>>>>>> serusers(a)lists.iptel.org >>>>>>> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] call forwarding rpid and billing
by Iqbal 07 Jun '05

07 Jun '05

Hi I am trying to make a call from pstn to IP which then forwards out to pstn again It all works fine, except the billing, which should be accounted in the IP account (although logically i cant see how this would work, since the IP user is not strictly speaking making the call) And the second problem is rpid, the rpid is showing the oriniating pstn caller, which is what it should, however if I wanted it to show the IP number instead (i.e the number assigned to that IP account) how could I. Iqbal

3 3

RE: [Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

Hi I added alias="blah.com" alias="sip.blah.com Tested and same result seen. if the callee hangs up only one BYE is noted in the acc table.. if the caller hangs up I note 3 BYE messages show in the acc table. I have 4 IP phones of different types and no matter what phone starts the call same issue is noted. Thanks for your help _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, 7 June 2005 5:10 PM To: Mr Greg Plater Subject: Re: [Serusers] Acc SetFlag issue It could be a loop, but I don't know where all the IPs are coming from. I cannot see alias="domain.com" in your ser.cfg. It's important to the myself check. You must be sure that you have ALL your possible domain names as aliases! g-)

3 5

Re: [Serusers] db_url not working
by Gorka Guridi 07 Jun '05

07 Jun '05

When you compiled SER, are you sure you had uncommented the following line: DEFS+=-DSQL_ACC in Makefile of acc? This file is in ser/modules/acc. If you wish to enable SQL accounting you must do it first.

1 0

Re: [Serusers] Acc SetFlag issue
by Greger V. Teigre 07 Jun '05

07 Jun '05

And you of course t_relay() BYE messages? (I don't think three BYEs in acc table has anything to do with where you put setflag. Either your UACs generate three BYEs because an ACK is not received and you don't t_relay, or something else is seriously wrong...) g-) ---- Original Message ---- From: Mr Greg Plater To: 'Greger V. Teigre' Sent: Tuesday, June 07, 2005 08:23 AM Subject: RE: [Serusers] Acc SetFlag issue > Hi > > I would normally agree with you in this should be simple. > > Tests show same result. when I make a call from one endpoint to > another and the callee hangs up, only a single BYE is noted in the > acc table. When the caller hangs up the call, I get three BYE > messages in the acc table. > > I checked for other setflag(1) and nothing is found. > > Any idea's > > > > > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Tuesday, 7 June 2005 4:00 PM > To: Mr Greg Plater; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Acc SetFlag issue > > This is a fairly straight-forward way to do it: > if (method=="INVITE" || method=="BYE" || method=="ACK") { > setflag(1); > }; > Put it at the top after sanity checks, but before loose routing etc. > g-) > > ---- Original Message ---- > From: Mr Greg Plater > To: serusers(a)lists.iptel.org > Sent: Monday, June 06, 2005 11:41 PM > Subject: [Serusers] Acc SetFlag issue > >> When I have setflag in this location a call logs three times. >> >> Were should I place the setflag if I wish for all calls to log in acc >> with mysql? >> >> >> Ser.cfg >> Snip: >> >> modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", 1) >> modparam("acc", "log_fmt", "cdfimorstup") >> modparam("acc", "db_flag", 1) >> modparam("acc", "log_missed_flag", 2) >> modparam("acc", "db_missed_flag", 2) >> modparam("acc", "report_cancels", 1) >> >> >> # ----------------------------------------------------------------- >> # Loose Route Section >> # >> ----------------------------------------------------------------- >> >> #setflag(1); >> >> if (loose_route()) { >> >> if (has_totag() && (method=="INVITE" || method=="ACK")) { >> >> if (client_nat_test("3")||search("^Route:.*;nat=yes")){ >> setflag(6); >> use_media_proxy(); >> }; >> }; >> >> route(1); >> break; >> }; >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] How to download SER
by Cecilia Muaddi 07 Jun '05

07 Jun '05

Hello, I just join the mail list and was hoping someone out there can point me to a location to download SER. It seems to require a user/password from the ftp site ftp://ftp.berlios.de/pub/ser. If so, how do I obtain the user/password. Thanks, Cecilia Muaddi cmuaddi(a)ambernetworks.net Amber Networks 408.626.7300 -------------------------------------------------- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. This E-mail is covered by the Electronic Communications Privacy Act, 18 U.S.C. §§ 2510-2521 and is legally privileged and/or confidential information. If you are not the intended recipient(s), or the employee or agent responsible for delivery of this message to the intended recipient(s), you are hereby notified that any dissemination, distribution or copying of this e-mail message is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete this e-mail message from your computer.

3 2

[Serusers] ACK not sent
by Greg Martin 07 Jun '05

07 Jun '05

I'm forwarding to a server which returns an invite with changed completion data. I've guarded against loops by assuring the returned invite isn't sent to the server again (I'm not sure that what I'm doing is compliant though). The correct phone rings and when I answer the dialed phone thinks it is connected. The rung phone never does think it's connected though and a packet capture shows no ack sent from the dialed phone which I presume is the problem (the phones are cisco 7960/7940). I'd appreciate any help. Sorry about the long message but felt the ngrep dump and cfg were both necessary. ********************************************************************************** debug=3 fork=yes log_stderror=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 1) modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method != "REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri != myself) { route(1); break; }; if (method == "REGISTER") { route(2); break; }; if (method == "INVITE") { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if(src_ip != "142.178.213.169" && src_ip != "142.178.213.201" && src_ip != "142.178.213.170" && src_ip != "142.178.213.202") { rewritehost("142.178.213.169"); forward(uri:host, uri:port); break; }; } route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; } *************************************************************************************** interface: qfe0 (142.178.218.0/255.255.254.0) filter: ip and ( port 5060 ) # U 142.178.21.214:50090 -> 142.178.218.210:5060 INVITE sip:5553331@142.178.218.210 SIP/2.0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:17 GMT. CSeq: 101 INVITE. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Content-Length: 249. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.169:5060 INVITE sip:5553331@142.178.213.169:5060;user=ip SIP/2.0. Max-Forwards: 10. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:17 GMT. CSeq: 101 INVITE. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Content-Length: 249. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. . # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. . # U 142.178.213.201:37403 -> 142.178.218.210:5060 INVITE sip:5552221@142.178.218.210:5060 SIP/2.0. Via: SIP/2.0/UDP 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Max-Forwards: 9. Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 249. Date: Mon, 06 Jun 2005 22:24:17 GMT. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531;received=142.178.213.201. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Server: Sip EXpress router (0.9.1 (sparc64/solaris)). Content-Length: 0. Warning: 392 142.178.218.210:5060 "Noisy feedback tells: pid=14006 req_src_ip=142.178.213.201 req_src_port=37403 in_uri=sip:5552221@142.178.218.210:5060 out_uri=sip:5552221@142.178.21.182:5060;user=ip via_cnt==3". . # U 142.178.218.210:5060 -> 142.178.21.182:5060 INVITE sip:5552221@142.178.21.182:5060;user=ip SIP/2.0. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Max-Forwards: 8. Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 249. Date: Mon, 06 Jun 2005 22:24:17 GMT. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52535 -> 142.178.218.210:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.21.182:52536 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. Date: Mon, 06 Jun 2005 22:24:23 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. . # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. Date: Mon, 06 Jun 2005 22:24:23 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. . # U 142.178.21.182:52537 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52538 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52539 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52540 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52541 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.214:50096 -> 142.178.218.210:5060 BYE sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on SIP/2.0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:31 GMT. CSeq: 102 BYE. User-Agent: CSCO/4. Content-Length: 0. Route: <sip:tc87sv01:5060;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>, <sip:5552221@142.178.21.182:5060>. . # Greg Martin Technology & Operations TELUS Communication Inc. office: 780-493-2786 cell: 780-718-4139

2 1

Re: [Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

Hi I would normally agree with you in this should be simple. Tests show same result. when I make a call from one endpoint to another and the callee hangs up, only a single BYE is noted in the acc table. When the caller hangs up the call, I get three BYE messages in the acc table. I checked for other setflag(1) and nothing is found. Any idea's _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, 7 June 2005 4:00 PM To: Mr Greg Plater; serusers(a)lists.iptel.org Subject: Re: [Serusers] Acc SetFlag issue This is a fairly straight-forward way to do it: if (method=="INVITE" || method=="BYE" || method=="ACK") { setflag(1); }; Put it at the top after sanity checks, but before loose routing etc. g-) ---- Original Message ---- From: Mr Greg Plater To: serusers(a)lists.iptel.org Sent: Monday, June 06, 2005 11:41 PM Subject: [Serusers] Acc SetFlag issue > When I have setflag in this location a call logs three times. > > Were should I place the setflag if I wish for all calls to log in acc > with mysql? > > > Ser.cfg > Snip: > > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "log_level", 1) > modparam("acc", "log_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "db_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "report_cancels", 1) > > > # ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > > #setflag(1); > > if (loose_route()) { > > if (has_totag() && (method=="INVITE" || method=="ACK")) { > > if (client_nat_test("3")||search("^Route:.*;nat=yes")){ > setflag(6); > use_media_proxy(); > }; > }; > > route(1); > break; > }; > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Registrering trouble. Perhaps Fifo
by Martin Kjeldsen 07 Jun '05

07 Jun '05

Hi there I'm having trouble registrering with my SER build (0.9.2), and using the auth_mysql ser.cfg from ONsip.org. It is as if SER isn't recieving any of the messages from my UA, and I've been wondering where I might have gone wrong. One thing I've found different from a former build (0.8.14), I have up and running is the socks which SER is listening on. When i do a serctl ps on the 0.9.2 i get this: ser@sipsite:~/running/ser0.9.0/sbin> serctl ps 200 ok 0 22531 attendant 1 22532 fifo server 2 22533 receiver child=0 sock= 212.97.222.xxx:5060 3 22534 receiver child=1 sock= 212.97.222.xxx:5060 4 22535 receiver child=2 sock= 212.97.222.xxx:5060 5 22536 receiver child=3 sock= 212.97.222.xxx:5060 6 22537 timer 7 22538 tcp receiver 8 22539 tcp receiver 9 22540 tcp receiver 10 22541 tcp receiver 11 22542 tcp main process And as far as I can see the same command on 0.8.14 yields almost the same output, except that the sock has a number like '0' and a @ sign is between scok and ip (eg: sock=0 @ ip-adress)! Can this be the cause of my troubles? I'm afraid that I'm not that much into Unix and Sockets! Any and all help will be greatly appreciated! Kind regards Martin

4 10

[Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

When I have setflag in this location a call logs three times. Were should I place the setflag if I wish for all calls to log in acc with mysql? Ser.cfg Snip: modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- #setflag(1); if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; };

2 1

[Serusers] Routing Logics based on MySQL Query !!!
by Felipe Martins 07 Jun '05

07 Jun '05

Hi everyone, I am planning to do all my routing logics, based on ser querying my MySQL in order to compare the prefix of the SURI and after that find the right route to perfectly route the call. I use SER 0.9.0 with Mysql Auth. Do anybody know how to do that ? This suggestion was given to me by Iqbal. Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

2 1

[Serusers] GrandStream BT 100 Voice Echo with SER
by Felipe Martins 07 Jun '05

07 Jun '05

Hi guys, I have a problem with my GrandStream Budge Tone 100. It's REGISTERed with my SER, I can make and receive calls with it but all the calls are voice echoed. Them same doesn't happen at the other softphone or hardphone I'm talking to. I mean, softphone calls to BT100, only BT100 remains with echo, the sofphone has excellent voice quality as bt100 but with no echo . Did anybody have the same problem ?? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

3 2

Re: [Serusers] Can I use realm name without DNS SRV
by Greger V. Teigre 07 Jun '05

07 Jun '05

SER will try to resolve using DNS SRV and A lookups unless you specify IP addresses, so: route[2] { #Host and port where Asterisk is listening, sip.conf rewritehostport("192.168.1.5:5070"); route(1); break; } g-) Antonio Cano damas wrote: > Hi, > > That seems to start working :). I've made the next changes: > listen="redstar.organization.org" > alias="192.168.1.5" > alias="organization.org" > alias="redstar.organization.org" > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > > And the proper updates into MySQL Database for domain like > 'redstar.organization.org'. > > Now I've got happening something strange, my UAs x-lite and > grandstream can register but Asterisk doesn't (at first look). > Looking the URIS I've seen that X-lite generate > @192.168.1.5redstar.organization.org while Asterisk generate > @redstar.organization.org. Finally I've got registering the Asterisk > using fromdomain= 192.168.1.5restar.organization.org :-/ > > My next step is try to make a call from X-Lite to one extension of > Asterisk. In the SER.cfg: > A) Into the INVITE process section: > if (uri =~ "^sip:(2[0-9][0-9])@*" ) { > log(1,"Al asterisk\n"); > route(2); > break; > }; > B) > route[2] > { > #Host and port where Asterisk is listening, sip.conf > rewritehostport("192.168.1.5redstar.organization.org:5070"); > route(1); > break; > } > > route[1] > { > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) { > #log(1, "Pasando por force_rtp_proxy\n"); > append_hf("P-Hint: Pasando por RTP_PROXY\r\n"); > force_rtp_proxy(); > }; > > # labeled all transaction for accounting > #setflag(4); > > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > } > > With that the INVITE petition gets an Request Timeout response, > making a little of ngrep: > > U IP_Client_UA:5060 -> 192.168.1.5:5060 > INVITE sip:213@192.168.1.5redstar.organization.org SIP/2.0..Via: > SIP/2.0/UDP > 192.168.0.6:5060;rport;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: > Anto > nio F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: > <sip:213@redstar.organization.org>..Contact: > <sip:10101@192.168.0.6:5060>..Call-ID: 4F13C7CB-D > 6AD-11D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 > INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: > X-Lite release 1103m..Content-Len > gth: 263....v=0..o=10101 5457996 5458244 IN IP4 > 192.168.0.6..s=X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 > RTP/AVP 3 0 8 98 101..a=rtpmap:0 pcmu/80 > 00..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 > iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15.. > # > > U 192.168.1.5:5060 -> IP_DNS_SERVER:5070 > INVITE sip:213@192.168.1.5redstar.organization.org:5070 > SIP/2.0..Record-Route: > <sip:213@192.168.1.5;ftag=1817289323;lr=on>..Via: SIP/2.0/UDP > 192.168.1.5;branc h=z9hG4bK151f.afc77ca4.0..Via: SIP/2.0/UDP > 192.168.0.6:5060;received=IP_Client_UA;rport=5060;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: > Antonio > F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: > <sip:213@redstar.organization.org>..Contact: > <sip:10101@IP_Client_UA:5060>..Call-ID: 4F13C7CB-D6AD-1 > 1D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 INVITE..Max-Forwards: > 69..Content-Type: application/sdp..User-Agent: X-Lite release > 1103m..Content-Length: > 283..P-Hint: Nat uac_test=3 ..P-Hint: Metodo INVITE corrigiendo > SDP..P-Hint: Pasando por RTP_PROXY....v=0..o=10101 5457996 5458244 IN > IP4 192.168.0.6..s= > X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 RTP/AVP 3 0 8 98 > 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 > gsm/8000..a=rtpmap:98 iLBC > /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15..a=direction:active.. > > Why SER sends the message to IP_DNS_SERVER in place of > 192.168.1.5redstar.organization.org? > > What's happen here? I don't understand nothing. What I make wrong? The > @192.168.1.5redstar.organization.org doesn't seem to be so elegant is > there any way to make it like redstar.organization.org > > Kind regards and thanks for your time, > Antonio F. Cano > > Greger V. Teigre wrote: > >> You can turn off SRV in X-Lite and Grandstream (Use DNS SRV option) >> And use SIP proxy: fully qualified domain name (same as outbound) >> g-) >> Antonio Cano damas wrote: >> >>> Hello all, >>> >>> First of all excuses for my bad english. I was looking into the list >>> and doesn't find nothing similar to my problem maybe I didn't use >>> the correct keywords. >>> >>> I've got a Proxy that is behaind and Public IP and I've got a >>> subdomain name pointing to this machine. But the DNS server isn't >>> mine and I can't add a SRV record, I've redirected all petitions to >>> this server and in that way if I make a petition of anykind service >>> (ssh, web, ...) to the subdomain.name.com it comes into the >>> machine. The problem is that all de UA i try (X-Lite, Grandstream and >>> Asterisk) makes DNS SRV queries and i don't know how to avoid this. >>> >>> For example: >>> >>> My subdomain is redstar.organization.org and I'd like to use the >>> realm organization.org >>> >>> + File /etc/hosts >>> 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org >>> redstar >>> >>> + In /etc/init.d/ser I've got this params line: >>> PARAMS="-P $PIDFILE -u root -g root -l >>> redstar.organization.org" --->I try using -l >>> organization.org, but the server said to me: 0(0) >>> ERROR: udp_init: bind(5, 0x80c8cc0, 16) on IP_Organization.org: >>> Cannot assign requested address >>> >>> + In /etc/ser/ser.cfg added this: >>> alias="organization.org" >>> alias="redstar.organization.org" >>> >>> + MySQL tables the domain field value is 'organization.org' >>> >>> When I start the SER it tells me: >>> Restarting ser: serListening on >>> Public_IP_Of_Redstar [192.168.1.5]:5060 >>> Aliases: redstar:5060 redstar.organization.org:* >>> organization.org:* The UA Grandstream config is: >>> SIP Server: organization.org >>> Outbound Proxy: redstar.organization.org >>> >>> And the received answer is a 478 Unresolveable destination. Can >>> anyone help me? What can I do? I'm a little desesperated :( >>> >>> Thanks in advance, >>> Antonio F. Cano >>> >>> Ngrep log result into redstar.organization.org machine: >>> >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> # >>> >>> Why this 'sip:192.168.1.5organization.org'? >>> >>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!.. >>> >>> >>> ###################################################################### >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> ######################################################### >>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>> >>> >>> om.B..@..........:..... >>> >>> # >>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!.. >>> >>> ########## >>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>> >>> >>> om.B..@..........:..... >>> >>> ################################################################################################################## >>> >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> ########################################################################################################################### >>> >>> U 192.168.1.5:5060 -> UA_Public_IP:62303 >>> SIP/2.0 478 Unresolveable destination (478/TM)..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083 >>> fbe2c60e48e32;rport=62303;received=IP_Client_UA..From: "Antonio F. >>> Cano (iGT)" <sip:10101@organization.org; >>> user=phone>;tag=b511698681c5146b..To: >>> <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 >>> f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: 100 >>> REGISTER..Server: Sip EXpress route >>> r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 >>> 80.38.244.122:5060 "Noisy feedback tells: >>> pid=8023 req_src_ip=IP_Client_UA req_src_port=62303 >>> in_uri=sip:192.168.1.5organization.org out_uri=sip:192. >>> 168.1.5organization.org >>> via_cnt==1".... >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Sip gateways
by Clay Bryan 07 Jun '05

07 Jun '05

I have recently found out that our current gateway that we use for our pstn will not do both h323 and sip at the same time. I am looking for a way to possible sip trunk from my Cisco Callmanger to ser or just use a cisco gateway to do it. I am not so sure about the cisco gateway supporting both h323 and sip. I would appreciate any advice on this situation. Thanks in advance. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

1 0

[Serusers] ldap+ser
by Alberto Aguilar 07 Jun '05

07 Jun '05

I heard somebody wanted to use LDAP instead of MySQL, currently Im developing a module to replace MySQL database with an LDAP directory. If you are interested please check the following URL: http://rocksteady.cs.fiu.edu/SER-LDAP/ Regards, Alberto

4 3

[Serusers] SIP_DOMAIN again
by Juan 06 Jun '05

06 Jun '05

Hi all. I just have my SER running on a FreeBSD machine. I put the export SIP_DOMAIN= 10.0.1.10 line in all the .profile and profile files I found but I still have the same "You need to set...."phrase when triyng to run serctl I also tried to do export from the console but I got export:command not found. The shell is csh. Any help will be appreciated. Juan Ferrari

2 2

[Serusers] configuration file
by Llanos Serna García-Conde 06 Jun '05

06 Jun '05

2 1

[Serusers] Sems-voicemail Vs Asterisk-voicemail
by Ahmed Ahmed 06 Jun '05

06 Jun '05

Hello all, I got SER working, to add voicemail service. I need a help which one is better and why Sems-voicemail or Asterisk-voicemail. Thanks Ahmed __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] can't open greeting
by Llanos Serna García-Conde 06 Jun '05

06 Jun '05

1 0

RE: [Serusers] Mediaproxy in LAN (?)
by Giudice, Salvatore 06 Jun '05

06 Jun '05

Mediaproxy will be between the clients for RTP only. Signaling will be relayed by SER. -----Original Message----- From: a cairo [mailto:a.cairo@scsitalia.it] Sent: Saturday, June 04, 2005 6:14 AM To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org Subject: RE: [Serusers] Mediaproxy in LAN (?) Hi all, i don't understand architecture of mediaproxy with ser ! Is mediaproxy between client and ser? Thanks -----Original Message----- From: "Giudice, Salvatore" <Salvatore.Giudice(a)FMR.COM> To: "bogdan aux" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> Date: Tue, 17 May 2005 20:10:46 -0400 Subject: RE: [Serusers] Mediaproxy in LAN (?) > I am trying to route a call from a non-routable 10.x.x.x network to a > phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to > SIP from my asterisk to an x-lite > > I found some references indicating that I should call: > > > if ( src_ip = IP_GATEWAY) > { > force_rport(); > fix_contact(); > use_media_proxy(); > }; > > Supposedly, you need the force_rport and the fix_contact to put media > proxy in the middle of the two rtp sessions, but I have et to see this > actually work. Has anyone been able to pass a call through media proxy > from a GW on a non-routable to a routable network, or even between two > ip's on the same network as the mediaproxy? > > If so, please share your configs. > > -----Original Message----- > From: bogdan aux [mailto:aux1d@yahoo.com] > Sent: Friday, May 13, 2005 6:16 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Mediaproxy in LAN (?) > > I want to route RTP packets between 2 computers in the > same network through Mediaproxy. I know this may have > no sense, but it's just a test configuration. > The call use_media_proxy() seems to have no effect, > the RTP packets go directly from one computer to > another. > My guess is mediaproxy makes a test on the IPs (they > are 192.168.147.2 and 192.168.147.3) and refuses to > proxy the call. > Is mediaproxy working just between different networks > or it's an error in my configuration file? > > > > > Yahoo! Mail > Stay connected, organized, and protected. Take the tour: > http://tour.mail.yahoo.com/mailtour.html > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] rewrite sip status
by Juan Priotti 06 Jun '05

06 Jun '05

Is it possible to rewrite a sip status received from a gateway? I mean, if I get a 183 Session Progress I want to rewrite it to a 180 Ringing. I've tried using the replace function but it only works for the "From" and "To" fiels. This is what I've used: onreply_route[1] { if (status =~ "183") { log(1, "183 received"); replace("183 Session Progress", "180 Ringing"); } Thank you Juan

1 0

[Serusers] Too Many Hops on Route :(
by Ozan Blotter 06 Jun '05

06 Jun '05

Dear Friends, This is fifth or tenth times writing to list but still no way :( REGISTERED USERS (12345 and 54321 are ATAs, 201 is Analog FXO's Port 0) [root@localhost ser]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'location' size : 512 table: 0xf50cb0f8 d_ll { n : 3 first: 0xf50cd108 last : 0xf50cd388 } ...Record(0xf50cd108)... domain: 'location' aor : '12345' ~~~Contact(0xf50cd168)~~~ domain : 'location' aor : '12345' Contact : 'sip:12345@192.168.1.200:5060' Expires : 33 q : Call-ID : '410D1B912D91111(a)192.168.1.200' CSeq : 7962 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50cd248)... domain: 'location' aor : '54321' ~~~Contact(0xf50cd2a8)~~~ domain : 'location' aor : '54321' Contact : 'sip:54321@192.168.1.201:5060' Expires : 34 q : Call-ID : '379D1B912DA1111(a)192.168.1.201' CSeq : 7955 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50cd388)... domain: 'location' aor : '201' ~~~Contact(0xf50cd3e8)~~~ domain : 'location' aor : '201' Contact : 'sip:201@192.168.1.35:5060' Expires : 37 q : Call-ID : 'c0a80123-13c4-3a570cf4-1f6-5973' CSeq : 184 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ---Domain--- name : 'aliases' size : 512 table: 0xf50c9068 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== (Basic SER.CFG , I want to it to route any 053XXXXXXXX Calls to 192.168.1.35 [Analog FXO Gateway] by adding 201 Prefix so it dials a Cellular. Nearly same thing worked for Brekeke's Ondo SIP Proxy but why id does not work in SER) debug=3 fork=yes log_stderror=no listen=192.168.1.10 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("12")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:053[0-9]*@.*") { prefix("201"); rewritehostport ("192.168.1.35:5060"); # forward ("192.168.1.35:5060"); # t_relay_to_udp("192.168.1.35","5060"); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; } SIPSAK RESULTS (Too Many Hops and User Not Found Errors) [root@localhost ser]# sipsak -T -s sip:20105353490056@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.098 ms) SIP/2.0 483 Too Many Hops 1: 192.168.1.10 (0.244 ms) SIP/2.0 404 User Not Found without Contact header [root@localhost ser]# sipsak -T -s sip:12345@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.084 ms) SIP/2.0 483 Too Many Hops 1: ?? (13.000 ms) SIP/2.0 200 OK Contact: <sip:12345@192.168.1.200:5060> [root@localhost ser]# sipsak -T -s sip:54321@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.079 ms) SIP/2.0 483 Too Many Hops 1: ?? (13.020 ms) SIP/2.0 200 OK Contact: <sip:54321@192.168.1.201:5060> [root@localhost ser]#

2 1

[Serusers] ruri rewrite question
by Medve Istvan 06 Jun '05

06 Jun '05

Hi All, if I rewrite ruri in the INVITE message I have to rewrite ruri in any subsequent messages (CANCEL, BYE, ACK) in the same dialog ? Or rewrite INVITE is enough ? thx, imedve

2 1

[Serusers] Can I use realm name without DNS SRV
by Antonio Cano damas 06 Jun '05

06 Jun '05

Hello all, First of all excuses for my bad english. I was looking into the list and doesn't find nothing similar to my problem maybe I didn't use the correct keywords. I've got a Proxy that is behaind and Public IP and I've got a subdomain name pointing to this machine. But the DNS server isn't mine and I can't add a SRV record, I've redirected all petitions to this server and in that way if I make a petition of anykind service (ssh, web, ...) to the subdomain.name.com it comes into the machine. The problem is that all de UA i try (X-Lite, Grandstream and Asterisk) makes DNS SRV queries and i don't know how to avoid this. For example: My subdomain is redstar.organization.org and I'd like to use the realm organization.org + File /etc/hosts 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org redstar + In /etc/init.d/ser I've got this params line: PARAMS="-P $PIDFILE -u root -g root -l redstar.organization.org" --->I try using -l organization.org, but the server said to me: 0(0) ERROR: udp_init: bind(5, 0x80c8cc0, 16) on IP_Organization.org: Cannot assign requested address + In /etc/ser/ser.cfg added this: alias="organization.org" alias="redstar.organization.org" + MySQL tables the domain field value is 'organization.org' When I start the SER it tells me: Restarting ser: serListening on Public_IP_Of_Redstar [192.168.1.5]:5060 Aliases: redstar:5060 redstar.organization.org:* organization.org:* The UA Grandstream config is: SIP Server: organization.org Outbound Proxy: redstar.organization.org And the received answer is a 478 Unresolveable destination. Can anyone help me? What can I do? I'm a little desesperated :( Thanks in advance, Antonio F. Cano Ngrep log result into redstar.organization.org machine: U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... # Why this 'sip:192.168.1.5organization.org'? U 192.168.1.5:32863 -> 195.235.113.3:53 Z............_sip._udp.192.168.1.5organization.org..!.. ###################################################################### U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... ######################################################### U 195.235.113.3:53 -> 192.168.1.5:32863 Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c om.B..@..........:..... # U 192.168.1.5:32863 -> 195.235.113.3:53 Z............_sip._udp.192.168.1.5organization.org..!.. ########## U 195.235.113.3:53 -> 192.168.1.5:32863 Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c om.B..@..........:..... ################################################################################################################## U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... ########################################################################################################################### U 192.168.1.5:5060 -> UA_Public_IP:62303 SIP/2.0 478 Unresolveable destination (478/TM)..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083 fbe2c60e48e32;rport=62303;received=80.32.90.5..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org; user=phone>;tag=b511698681c5146b..To: <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Server: Sip EXpress route r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 80.38.244.122:5060 "Noisy feedback tells: pid=8023 req_src_ip=80.32.90.5 req_src_port=62303 in_uri=sip:192.168.1.5organization.org out_uri=sip:192. 168.1.5organization.org via_cnt==1"....

2 1

[Serusers] Problem with installing SER. Howto SER Step by Step?
by MapsAir 06 Jun '05

06 Jun '05

Dear Helpers! I am very new to linux and SER. I have read the article on your website and am very interested in learning about the sip proxy SER. However, I am still unable to install the SER. Would you please tell me how to install the SER or directly me to an article of step by step how to install the SER. I have read you how to. However, I'm pretty now to linux, and I am still unable to install it. I would be very appreciated for your time! Thanks in advance! Lan Dear Helpers! I am very new to linux and SER. I have read the article on your website and am very interested in learning about the sip proxy SER. However, I am still unable to install the SER. Would you please tell me how to install the SER or directly me to an article of step by step how to install the SER. I have read you how to. However, I'm pretty now to linux, and I am still unable to install it. I would be very appreciated for your time! Thanks in advance! Lan

3 2

[Serusers] rpid again
by Dirk Willbrandt 06 Jun '05

06 Jun '05

Hi list ! I have a Problem with the appending of an rpid with the functions avp_load_radius(), append_rpid_hf(). In my Radius DB i have an SIP-AVP field which is filled with the Value rpid:[NUMBER] When i run manuall the Radius Reply-Query i get the correct value for SIP-AVP. Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed after an incoming INVITE. My Radius is answering on the command and sends the correct SIP-AVP Value for the User. Now my Problem: I can't append the RPID or anything else to the SIP-Header after the answer from the Radius. So the call is going out without the rpid in the Header. Can anyone help me ? Here is the Configure-Snip: if (method=="INVITE") { # here comes the invite if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it seems that i must use an proxy auth proxy_challenge("","0"); # and an proxy challenge if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the radius log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a message break; # go out } avp_print(); # print all avp's (this command is never executed :( ) append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row - it is never executed append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); # append the rpid from SIP-AVP - and this command is also never executed break; # go out }; route(3); # go to route block 3 break; # end } I tried also this snippet without the breaks and proxy-commands but it doesn't work. I don't know what is wrong - please help me :s Thanks ! Dirk

3 2

Re: [Serusers] ser don't start nedd help
by harry gaillac 06 Jun '05

06 Jun '05

I drop create ser db. However when I start ser i get : serveur1:~# tail /var/log/syslog Jun 3 18:39:01 serveur1 /USR/SBIN/CRON[2616]: (root) CMD ( [ -d /var/lib/php4 ] && find /var/lib/php4/ -type f -cmin +$(/usr/lib/php4/maxlifetime) -print0 | xargs -r -0 rm) Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Maxfwd module- initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: permissions - initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: WARNING: File not found: /etc/ser/permissions.allow Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: WARNING: File not found: /etc/ser/permissions.deny Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: AVPops - initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: ERROR: fix_expr : fix_actions error version table from phpmyadmin: Résultat de la requête SQL Serveur: localhost Base de données: ser Généré le : Vendredi 03 Juin 2005 à 18:39 Généré par: phpMyAdmin 2.6.2 / MySQL 4.0.24_Debian-10-log requête SQL: SELECT * FROM `version` LIMIT 0, 30; Enregistrements: 27 table_name table_version subscriber 5 reserved 1 phonebook 1 pending 4 missed_calls 2 location 6 grp 2 event 1 aliases 6 active_sessions 1 acc 2 config 1 silo 3 realm 1 domain 1 uri 1 server_monitoring 1 server_monitoring_agg 1 trusted 1 usr_preferences 2 usr_preferences_types 1 admin_privileges 1 calls_forwarding 1 speed_dial 2 gw 1 gw_grp 1 lcr 1 Version table with command: +-----------------------+---------------+ | table_name | table_version | +-----------------------+---------------+ | subscriber | 5 | | reserved | 1 | | phonebook | 1 | | pending | 4 | | missed_calls | 2 | | location | 6 | | grp | 2 | | event | 1 | | aliases | 6 | | active_sessions | 1 | | acc | 2 | | config | 1 | | silo | 3 | | realm | 1 | | domain | 1 | | uri | 1 | | server_monitoring | 1 | | server_monitoring_agg | 1 | | trusted | 1 | | usr_preferences | 2 | | usr_preferences_types | 1 | | admin_privileges | 1 | | calls_forwarding | 1 | | speed_dial | 2 | | gw | 1 | | gw_grp | 1 | | lcr | 1 | +-----------------------+---------------+ 27 rows in set (0.01 sec) --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > they should be identical, do u keep logs of queries > sent to Mysql , if > so look at them when u use phpmyadmin, and also > manual queries, and you > will see what is actually being sent, the log will > usually be in > /var/log on the machine with the DB > > Iqbal > > harry gaillac wrote: > > >Why datas from phpmyadmin are not equal to > >mysql> select * from version; > > > >Is it a bug ? > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>whats the error, I cut and pasted that query and > it > >>deleted the row, > >> > >>Iqbal > >> > >>harry gaillac wrote: > >> > >> > >> > >>>Do you know the right SQL command to delete > >>> > >>> > >>multiple > >> > >> > >>>values ? > >>>I try > >>>mysql> delete from version where > >>> > >>> > >>table_name='trusted'; > >> > >> > >>>it's not good > >>> > >>>Harry > >>> > >>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>> > >>> > >>> > >>> > >>> > >>>>what does select * from version display > >>>> > >>>>Iqbal > >>>> > >>>>harry gaillac wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Ok, > >>>>> > >>>>>I understand . > >>>>>I use phpmyadmin to manage ser however > >>>>>mysql> select * from version; don't display the > >>>>> > >>>>> > >>>>> > >>>>> > >>>>same > >>>> > >>>> > >>>> > >>>> > >>>>>result > >>>>> > >>>>>Harry > >>>>> > >>>>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>now remove all the duplicate rows, leaving > only > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>one > >>>> > >>>> > >>>> > >>>> > >>>>>>of each type, eg if > >>>>>>you have alook there are two rows called > >>>>>>domains...remove one, lcr is > >>>>>>also there twice, > >>>>>> > >>>>>>Iqbal > >>>>>> > >>>>>>harry gaillac wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>here is table version from ser DB: > >>>>>>>+-----------------------+---------------+ > >>>>>>>| subscriber | 5 | > >>>>>>>| reserved | 1 | > >>>>>>>| phonebook | 1 | > >>>>>>>| pending | 4 | > >>>>>>>| missed_calls | 2 | > >>>>>>>| location | 6 | > >>>>>>>| grp | 2 | > >>>>>>>| event | 1 | > >>>>>>>| aliases | 6 | > >>>>>>>| active_sessions | 1 | > >>>>>>>| acc | 2 | > >>>>>>>| config | 1 | > >>>>>>>| silo | 3 | > >>>>>>>| realm | 1 | > >>>>>>>| domain | 1 | > >>>>>>>| uri | 1 | > >>>>>>>| server_monitoring | 1 | > >>>>>>>| server_monitoring_agg | 1 | > >>>>>>>| trusted | 1 | > >>>>>>>| usr_preferences | 2 | > >>>>>>>| usr_preferences_types | 1 | > >>>>>>>| admin_privileges | 1 | > >>>>>>>| calls_forwarding | 1 | > >>>>>>>| speed_dial | 2 | > >>>>>>>| gw | 1 | > >>>>>>>| gw_grp | 1 | > >>>>>>>| lcr | 1 | > >>>>>>>| subscriber | 5 | > >>>>>>>| reserved | 1 | > >>>>>>>| phonebook | 1 | > >>>>>>>| pending | 4 | > >>>>>>>| missed_calls | 2 | > >>>>>>>| location | 6 | > >>>>>>>| grp | 2 | > >>>>>>>| event | 1 | > >>>>>>>| aliases | 6 | > >>>>>>>| active_sessions | 1 | > >>>>>>>| acc | 2 | > >>>>>>>| config | 1 | > >>>>>>>| silo | 3 | > >>>>>>>| domain | 1 | > >>>>>>>| domain | 1 | > >>>>>>>| uri | 1 | > >>>>>>>| server_monitoring | 1 | > >>>>>>>| server_monitoring_agg | 1 | > >>>>>>>| trusted | 1 | > >>>>>>>| usr_preferences | 2 | > >>>>>>>| usr_preferences_types | 1 | > >>>>>>>| admin_privileges | 1 | > >>>>>>>| calls_forwarding | 1 | > >>>>>>>| speed_dial | 2 | > >>>>>>>| gw | 1 | > >>>>>>>| gw_grp | 1 | > >>>>>>>| lcr | 1 | > >>>>>>>+-----------------------+---------------+ > >>>>>>>54 rows in set (0.00 sec) > >>>>>>> > >>>>>>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>>can u dump your version table here, and let > us > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>look > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>>at it > === message truncated === _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

4 13

[Serusers] SER 0.9.2?
by Dennis Real 06 Jun '05

06 Jun '05

Hi, where can I get SER Version 0.9.2? What switch do I need for cvs? regards

2 1

[Serusers] RE: Re: radiusclient
by Kamran Ahmad 06 Jun '05

06 Jun '05

any comments how to reslove this issue auth_radius.so>: /usr/local/lib/ser/modules/auth_radius.so: undefined symbol: fm_malloc May 26 16:32:04 voice ser: parse error (25,13-14): failed to load module May 26 16:32:04 voice ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/uri_radius.so>: /usr/local/lib/ser/modules/uri_radius.so: undefined symbol: fm_malloc May 26 16:32:04 voice ser: parse error (35,13-14): failed to load module May 26 16:32:04 voice ser: set_mod_param_regex: No module matching auth_radius found | May 26 16:32:04 voice ser: parse error (48,18-19): Can't set module parameter May 26 16:32:04 voice ser: set_mod_param_regex: No module matching auth_radius found | May 26 16:32:04 voice ser: parse error (49,18-19): Can't set module parameter __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

2 1

[Serusers] 0.9.0 version status?
by Jose Bertuzzi 06 Jun '05

06 Jun '05

Hello, is version 0.9.0 stable? Do you recommend to install it, or 0.8.14 works better? I am running 0.8.14 compiled from source on freebsd. I had some trouble trying to compile radius modules so I think 0.9.0 could fix this. thanks for your recommendation, Pablo. --------------------------------- Discover Yahoo! Use Yahoo! to plan a weekend, have fun online & more. Check it out!

2 1

[Serusers] (no subject)
by news 06 Jun '05

06 Jun '05

Hello seruser, I'm using SER 0.8.14 with freeradius 1.0.2 and radiusclient 0.4.8 . All connections are going through a rtpproxy. Now my Question: Is it possible to get a "watchdog" - functionality with SER? I need to get an entry (stop or failure) in my detail-yyyy-mm-dd file when the connection has been terminated by an error or by disconnecting one phone from the net. thanks -- ___________________________ Stefan Abu Salah Stefan(a)AbuSalah.com www.AbuSalah.com 0179 140 89 17

1 0

[serusers] logs
by michael p 06 Jun '05

06 Jun '05

hi where can i find the logs errors and others of ser? thanks M.

2 2

[Serusers] SER + Asterisk
by Siphiwo Dzingwe 06 Jun '05

06 Jun '05

Hi I have recently installed SER 0.8.14 and I am trying to make it talk to asterisk. How do I do that?

1 0

[Serusers] Re: [Serdev] UA request responses that do not conform RFC3261 nor RFC2543
by Jan Janak 06 Jun '05

06 Jun '05

The best way would be to fix the user agent. SER would not know where to forward the reply if it does not contain Via. Jan. On 02-06-2005 10:58, Dominik Schulte wrote: > Hi, > > I am struggling with a problem where I have a UA that responds to invites but > does not preserve the original request Via fields. see ngrep capture below. > I can see that SER receives the reply, but drops it because it cannot correlate any transaction with this response. > I tried to write a module that uses the tm modules callbacks, but it seems that response messages are dropped by the core. > > Is there any way I can validate responses that fall outside any transaction?? > > # > U 172.17.26.19:5060 -> 172.17.26.178:5060 > INVITE sip:dschulte@heat.callsuite.tellumat.co.za SIP/2.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:dschulte@heat.callsuite.tellumat.co.za>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 70. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > > # > U 172.17.26.178:5060 -> 172.17.26.19:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:dschulte@heat.callsuite.tellumat.co.za>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Server: Sip EXpress router (0.8.14 (i386/linux)). > Content-Length: 0. > Warning: 392 172.17.26.178:5060 "Noisy feedback tells: pid=17762 req_src_ip=172.17.26.19 req_src_port=5060 in_uri=sip:dschulte@heat.callsuite.tellumat.co.za out_uri=sip:94099:1001@172.17.1.31 via_cnt==1". > . > > # > U 172.17.26.178:5060 -> 172.17.1.31:5060 > INVITE sip:94099:1001@172.17.1.31 SIP/2.0. > Record-Route: <sip:dschulte@172.17.26.178;ftag=859273871;lr=on>. > Via: SIP/2.0/UDP 172.17.26.178;branch=z9hG4bK33ab.6f0be065.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 69. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > # > U 172.17.1.31:5060 -> 172.17.26.178:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 172.17.26.178. > `From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>;tag=viper94099. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Content-Length: 0. > . > > > # > U 172.17.1.31:5060 -> 172.17.26.178:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 172.17.26.178. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: "" <sip:94099@172.17.1.31>;tag=viper94099. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Content-Length: 0. > . > > > > > # > U 172.17.26.178:5060 -> 172.17.1.31:5060 > INVITE sip:94099:1001@172.17.1.31 SIP/2.0. > Record-Route: <sip:dschulte@172.17.26.178;ftag=859273871;lr=on>. > Via: SIP/2.0/UDP 172.17.26.178;branch=z9hG4bK33ab.6f0be065.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 69. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > > > ********************************************************************** > Relevant company disclaimers are available at the following addresses: > Tellumat (Pty) Ltd e-mail: mailto:disclaimer@tellumat.com?Subject=Tellumat_Disclaimer > Web: http://www.tellumat.com/email.aspx > ********************************************************************** > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev

1 0

[Serusers] pb with jabber
by mma 06 Jun '05

06 Jun '05

i add the following entries on the sip_jab.jusers table insert into jusers values ('','user1jabber','password','sip:user1@sip.server.net',0); insert into jusers values ('','user2jabber','password','sip:user1@sip.server.net',0); i created user1jabber and user2jabber using exodus as jabber client i created also user1 and user2 on ser whiwc is running on 192.168.1.70 (sip.server.net) jabber is runninng also on the same machine : 192.168.1.70 (jabber.server.net) jabber works fine, i have tested it; DNS server : 195.167.203.168 i want the two users to communicate using IM user1 is connecting to windows messenger : user1(a)sip.server.net user2 is connecting to exodus : user2jabber(a)jabber.server.net user1 send an IM to user2jabber using user2jabber*jabber.org(a)jabber.server.net the problems are : -user2jabber don't receive what user1 sends. -also pa modules does not work : i have no response from the server when a SUBSCRIBE request is sent (i added user3 as a contact to user1 ) please i need help regards ///////////////////////////////////////////////////////////////////////////////////// No. Time Source Destination Protocol Info 25 12.150821 192.168.1.118 195.167.203.168 DNS Standard query A sip.server.net 26 12.153145 195.167.203.168 192.168.1.118 DNS Standard query response A 192.168.1.70 27 12.160197 192.168.1.118 192.168.1.70 SIP Request: REGISTER sip:sip.server.net 28 12.164436 192.168.1.70 192.168.1.118 SIP Status: 200 OK (1 bindings) 29 12.166088 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted (1 bindings) 30 12.169036 192.168.1.118 192.168.1.70 SIP Request: SUBSCRIBE sip:user3@sip.server.net 31 12.172514 192.168.1.70 192.168.1.118 SIP Status: 400 Bad Request ..... 171 21.841023 192.168.1.118 192.168.1.70 SIP/SDP Request: INVITE sip:user2jabber*jabber.org@jabber.server.net, with session description 172 21.843627 192.168.1.70 192.168.1.118 SIP Status: 202 Accepted 173 21.861278 192.168.1.118 192.168.1.70 SIP Request: ACK sip:user2jabber*jabber.org@jabber.server.net 174 21.862529 192.168.1.118 192.168.1.70 SIP Request: MESSAGE sip:user2abber*jabber.org@jabber.server.net (text/plain) 175 21.865357 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted ..... 297 22.446316 192.168.1.118 192.168.1.70 SIP Request: BYE sip:user2jabber*jabber.org@jabber.server.net

1 2

[Serusers] pb with jabber
by mma 06 Jun '05

06 Jun '05

i add the following entries on the sip_jab.jusers table insert into jusers values ('','user1jabber','password','sip:user1@sip.server.net',0); insert into jusers values ('','user2jabber','password','sip:user1@sip.server.net',0); i created user1jabber and user2jabber using exodus as jabber client i created also user1 and user2 on ser whiwc is running on 192.168.1.70 (sip.server.net) jabber is runninng also on the same machine : 192.168.1.70 (jabber.server.net) jabber works fine, i have tested it; DNS server : 195.167.203.168 i want the two users to communicate using IM user1 is connecting to windows messenger : user1(a)sip.server.net user2 is connecting to exodus : user2jabber(a)jabber.server.net user1 send an IM to user2jabber using user2jabber*jabber.org(a)jabber.server.net the problems are : -user2jabber don't receive what user1 sends. -also pa modules does not work : i have no response from the server when a SUBSCRIBE request is sent (i added user3 as a contact to user1 ) please i need help regards ///////////////////////////////////////////////////////////////////////////////////// No. Time Source Destination Protocol Info 25 12.150821 192.168.1.118 195.167.203.168 DNS Standard query A sip.server.net 26 12.153145 195.167.203.168 192.168.1.118 DNS Standard query response A 192.168.1.70 27 12.160197 192.168.1.118 192.168.1.70 SIP Request: REGISTER sip:sip.server.net 28 12.164436 192.168.1.70 192.168.1.118 SIP Status: 200 OK (1 bindings) 29 12.166088 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted (1 bindings) 30 12.169036 192.168.1.118 192.168.1.70 SIP Request: SUBSCRIBE sip:user3@sip.server.net 31 12.172514 192.168.1.70 192.168.1.118 SIP Status: 400 Bad Request ..... 171 21.841023 192.168.1.118 192.168.1.70 SIP/SDP Request: INVITE sip:user2jabber*jabber.org@jabber.server.net, with session description 172 21.843627 192.168.1.70 192.168.1.118 SIP Status: 202 Accepted 173 21.861278 192.168.1.118 192.168.1.70 SIP Request: ACK sip:user2jabber*jabber.org@jabber.server.net 174 21.862529 192.168.1.118 192.168.1.70 SIP Request: MESSAGE sip:user2abber*jabber.org@jabber.server.net (text/plain) 175 21.865357 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted ..... 297 22.446316 192.168.1.118 192.168.1.70 SIP Request: BYE sip:user2jabber*jabber.org@jabber.server.net ////////////////////////////////////////////////////////////////////////////////////// ser.cfg ***************** debug=4 # debug level (cmd line: -dddddddddd) #fork=yes # (cmd. line: -D) fork=no log_stderror=yes # (cmd line: -E) #log_stderror=no # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 listen=sip.server.net listen=192.168.1.70 fifo="/tmp/ser_fifo" #fifo="/tmp/ser_fifoql://ser:heslo@localhost/ser" # for more info: sip_router -h # ------------------ module loading ---------------------------------- #modules loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- # -- registrar params -- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) # -- usrloc params -- modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_mode", 0) # -- jabber params -- modparam("jabber","db_url","mysql://jabber:jabber@127.0.0.1/sip_jab") modparam("jabber","jaddress","jabber.server.net") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.server.net=*") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; if (uri=~"[@:]sip\.server\.net([;:].*)*") #if (uri=~"^sip:(.+@)?(192\.168\.1\.70|(jabber\.)?server\.net)([:;\?].*)?$") { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { log("XJAB:methode SUBSCRIBE \n"); if(jab_go_offline()) if (t_newtran()) { log("XJAB: subscribe registrar\n"); # subscribe("registrar"); handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; }; if (search("To:.*@jabber\.server\.net")) { log("XJAB : search for jabber.sofiphone.net /n"); if (! t_newtran()) { sl_reply_error(); break; }; if (method=="MESSAGE") { log("MESSAGE received -> manage it with XJAB\n"); if(search("\n:on")) { if (jab_go_online()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:off")) { if (jab_go_offline()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:join")) { if (jab_join_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if(search("\n:exit")) { if (jab_exit_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if (jab_send_message()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("503","Service Unavailable"); }; break; }; if (method=="SUBSCRIBE") { log("subscribe jabber /n"); # subscribe("jabber"); handle_subscription("jabber"); break; }; log("NON_Message request received for JABBER gateway->dropt!\n"); sl_send_reply("202","Accepted"); break; }; if (!t_relay()) { sl_reply_error(); }; #forward(uri:host,uri:port); } /////////////////////////////////////////////////////////////////////////////////////

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[Serusers] SERWEB and MySQL Connection ERROR Database error: pconnect(192.168.1.246, ser, $Password) failed.
by RockV 05 Jun '05

05 Jun '05

Hi, I am not an expert, but I think I have found a solution. First, you should point serweb to your mysql database. //////////////////////////////////////////////////////////////// // configure database /* these are the defaults with which SER installs; if you changed the SER account for MySQl, you need to update here */ $this->db_host="127.0.0.1"; //database host $this->db_name="ser"; //database name $this->db_user="ser"; //database conection user $this->db_pass="heslo"; //database conection password In my case localhost but if I set localhost in $this->db_host it doesn't work. So, you must set IP Address... Then, you should check if you can enter to MySQL with IP Address. In my case: mysql -h 127.0.0.1 -u ser -pheslo ser Is not enough if you can enter with: mysql -h localhost -u ser -pheslo ser I think that will help. RockV >yup password error :-), can you connect to the db from the linux command >line > >mysql -h a.b.c.d -u username -ppassword ser > >do this from the machine serweb is hosted on. > >Also I would avoid serweb until you have ser up and running properly, >because serweb comes with its own little headaches :-) > >Iqbal > >Manoj Ghag wrote: > >> hi, >> In case of MYSQL .. I am getting following ERROR when I try to open a >> SERWEB.... >> >> >> *Warning*: mysql_pconnect(): Access denied for user: 'ser at test' (Using >> password: YES) in */var/www/phplib/db_mysql.inc* on line *73* >> *Database error:* pconnect(192.168.1.246, ser, $Password) failed. >> *MySQL Error*: () >> Session halted. >> >> >> ANY HELP ? >> >> >> ------------------------------------------------------------------------ >> Yahoo! Mail >> Stay connected, organized, and protected. Take the tour >> <HYPERLINK "http://tour.mail.yahoo.com/mailtour.html"http://tour.mail.yahoo.com/mailtou r.html> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Serusers mailing list >>Serusers at iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 04/06/2005

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[Serusers] need help to route calls to PSTN
by harry gaillac 05 Jun '05

05 Jun '05

Hello ser users, I configure ser and asterisk on the same box (+nat). I can send and receive calls from my domain or other domains however i can't send uri to pstn : ser listen on port 5060 / public ip asterisk listen 5040 / private address 192.168.0.50 When I dial a number to pstn phone send sip:1234@mydomain:5060 but ser don't rewrite host port according to ser.cfg . What's wrong ? Why ? # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.9.253 # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # --domain params -- modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") { record_route(); }; if (method=="ACK" || method=="BYE") { unforce_rtp_proxy(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # Requests to others domains if (!is_from_local()) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (is_from_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (!search("^Contact:") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("19")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (!is_from_local()) { route(1); break; }; if (src_ip==192.168.0.50 && allow_trusted()) { route(4); route(1); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; break; }; if (uri=~"sip:^1013@.*$" || uri=~"sip:^[0-9]{10}@.*$" || uri=~"sip:^06[0-9]{8}@.*$") { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); break; } route[4] { ########################## # Incoming PSTN Handler # ########################## if (uri=~"^[0-9]{10}@.*$") { strip(8); break; }; } route[5] { ########################## # Outgoing PSTN Handler # ########################## if (uri=~"sip:^1013@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^06[0-9]{8}@.*$") { if (!is_user_in("Request-URI", "cell phone")) { sl_send_reply("403", "Forbidden"); break; }; rewritehostport("192.168.50:5040"); }; } onreply_route[1] { if (isflagset(6) && status=~"(180) | (183) | 2[0-9][0-9]") { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

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[Serusers] Early media
by Olivier Taylor 05 Jun '05

05 Jun '05

Hi guys, We use ser, rtpproxy and sems. It seems that some gateways(Cisco) doesn't send the Ringing, only the 100 Trying. Is there a way to force UA to accept early media? It seems that's some UA doesn't accept that early media (Grandstream and Sipura), Eyebeam form Xten works correctly... Any ideas? Olivier -- Olivier Taylor

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[Serusers] SER + SEMS Voicemail Feature
by Narcis GRATIANU 04 Jun '05

04 Jun '05

Hello ! I am using SER with SEMS for voicemail, but, for now, the voicemail is active only when the user is not authenticated. Does anybody has a config sample for the voicemail to kick in after, let's say, few rings, also when the user is authenticated ? Thank you !

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[Serusers] routing to PSTN
by harry gaillac 04 Jun '05

04 Jun '05

Hello, Here is my ser.cfg I try to send uri not in location or aliase table . INVITE message is not sent to gateway just 404 is returned ? What's wrong ? Harry # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.10.96 alias="nxs.yi.org" # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- nathelper params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") { record_route(); }; if (method=="ACK" || method=="BYE") { unforce_rtp_proxy(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # Requests to others domains if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (!search("^Contact:") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("19")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (!uri==myself) { route(1); break; }; if (src_ip==192.168.0.50 && allow_trusted()) { route(4); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; break; }; if (uri=~"sip:^[0-9]{2}@.*$" || uri=~"sip:^1013@.*$" ||uri=~"sip:^[0-9]{10}@.*$" || uri=~"sip:^06[0-9]{8}@.*$") { route(5); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; t_on_reply("1"); if (!t_relay()) { if (isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[4] { ########################## # Incoming PSTN Handler # ########################## if (uri=~"^[0-9]{10}@.*$") { strip(8); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[5] { ########################## # Outgoing PSTN Handler # ########################## if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "internal")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^1013@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^06[0-9]{8}@.*$") { if (!is_user_in("Request-URI", "cell phone")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("192.168.0.50:5040"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180) | (183) | 2[0-9][0-9]") { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] billing question (some architectural case)
by Wojciech Ziniewicz 04 Jun '05

04 Jun '05

Hello, I 've just came to the point of rolling my own billing that I have mysql database with table "acc" and details of called party , accstart, acc period and so on. I have another table that I have prefixes and prices for T1,T2,T3 . My question is : do You have some howtos, scripts for rating users? I try bash for now but it is slow and not scalable in future.. what about mysql shell or sth? Could you send me some examples ? I would appreciate that Regards..... -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

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[Serusers] ser don't start
by harry gaillac 04 Jun '05

04 Jun '05

Hello ser users, I've sent many mails about starting ser. my problem is solved . ser sent to syslog: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) auth_db module did not find table "subsciber" below. if (!www_authorize("nxs.yi.org", "subsciber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; I fix table to "subscriber" ! Thanks ser users for help Regards Harry _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

RE: [Serusers] Mediaproxy in LAN (?)
by Giudice, Salvatore 04 Jun '05

04 Jun '05

I am trying to route a call from a non-routable 10.x.x.x network to a phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to SIP from my asterisk to an x-lite I found some references indicating that I should call: if ( src_ip = IP_GATEWAY) { force_rport(); fix_contact(); use_media_proxy(); }; Supposedly, you need the force_rport and the fix_contact to put media proxy in the middle of the two rtp sessions, but I have et to see this actually work. Has anyone been able to pass a call through media proxy from a GW on a non-routable to a routable network, or even between two ip's on the same network as the mediaproxy? If so, please share your configs. -----Original Message----- From: bogdan aux [mailto:aux1d@yahoo.com] Sent: Friday, May 13, 2005 6:16 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Mediaproxy in LAN (?) I want to route RTP packets between 2 computers in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another. My guess is mediaproxy makes a test on the IPs (they are 192.168.147.2 and 192.168.147.3) and refuses to proxy the call. Is mediaproxy working just between different networks or it's an error in my configuration file? Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

4 3

[Serusers] CDRTool
by Mr Greg Plater 03 Jun '05

03 Jun '05

Is anyone using the CDRTool from AG I have installed and configured it for testing. Its missing the Rating engine etc. Anyone know the cost of a Lic with the rating engine ?

2 2

[Serusers] Asterisk and SER
by Ronald Wiplinger 03 Jun '05

03 Jun '05

I read about SER and I have installed Asterisk on one machine. Unfortunately this machine is broken and the replacement will take longer as I wished. I have a P-4, 1GHz with 1 GB RAM, which was planned to be used as SER anyway. I would like to use this downtime of the Asterisk box to install SER. I understand that SER and Asterisk can work together, while SER makes connection, Asterisk will add features. How do I do that efficiently? 1. SER needs a public IP, does then Asterisk need a pulic IP, or can it just be on the internal LAN? 2. Now the password for each phone is made in the SIP and IAX config files. How can I do that than in SER? If I use reatime, can I use the same database? 3. A user can make a phone call to the users on our system (Asterisk domain) and via gateways. We use ASTCC for billing. How can I use this together with SER. I am not sure if I understand it correct, but would for internal calls (calls without features) be handeled directly from SER, while if the user need a feature, than he would be transfered to Asterisk? How can I setup multiple Asterisk boxes with one SER, but all Asterisk machines share the same dial plan (realtime) I wanted to read more first, but the down time of my * forces me to do a SER setup faster, so that my users can quickly (by office hours Monday) at least make internal calls. Any hint is welcome ;-) bye Ronald

3 2

[Serusers] SER 0.9.2 & Route & Analog FXO Problems :(
by Ozan Blotter 03 Jun '05

03 Jun '05

Dear Friends, I know this is annoying to post nearly same things but still no answer :( Is it really so hard or can anybody tell me why this is not working. There are : ATA 1 : 12345 (registered and can call 54321, no problem) ATA 2 : 54321 (registered and can call 12345, no problem) ANALOG FXO : 201 (registered as 201, for test purposes i only registered one port of it) With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it worked, even i can add more prefixes no problem. But in SER i cannot :( I'm using ver 0.9.2. Normally it's like when i want to call a GSM, proxy simply adds a prefix 201 which is same with registered sip number for regarding analog port and forwards it (looks like sip:20105353490056@192.168.1.35) to Gateway and Gateway simply strips 201 from it and dials 05353490056, yes it connects. But what this same does not work on SER, any ideas because it's very boring to play all day and night ? Thanks.... >>>>>>>>>> minimal SER.CFG >>>>>>>>>> debug=3 fork=yes log_stderror=no check_via=no dns=no rev_dns=no listen=192.168.1.10 port=5060 children=4 fifo="/tmp/ser_fifo" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; record_route(); if (loose_route()) { t_relay(); break; }; if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; if (uri=~"^sip:053[0-9]*@.*") { prefix("201"); rewritehostport ("192.168.1.35:5060"); # forward ("192.168.1.35:5060"); t_relay_to_udp("192.168.1.35","5060"); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (!t_relay()) { sl_reply_error(); }; } >>>>>>>>>> call from 12345 to 54321 (SIP 2 SIP Call) >>>>>>>>>> U 192.168.1.10:5060 -> 192.168.1.201:5060 INVITE sip:54321@192.168.1.201:5060 SIP/2.0..Record-Route: <sip:192.168.1.1 0;ftag=xITO2cDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9hG4bK233f .8f8b9d41.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKba4b3e1d3eed 9c7c..Max-Forwards: 16..To: <sip:54321@192.168.1.10>..From: <sip:12345@192. 168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID: 911D1B912D91212(a)192.168.1.20 0..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..Session-Expires : 100;refresher=uac..Supported: timer..Content-Type: application/sdp..Conte nt-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-..c=IN IP4 192. 168.1.200..t=0 0..m=audio 2142 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=send recv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bKba4b 3e1d3eed9c7c..To: <sip:54321@192.168.1.10>;tag=xETOxUDMxED..From: <sip:1234 5(a)192.168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID: 911D1B912D91212(a)192.16 8.1.200..CSeq: 1 INVITE..Record-Route: <sip:192.168.1.10;ftag=xITO2cDMxID;l r=on>..Content-Length: 0.... >>>>>>>>>> call from 12345 to 05353490056 (GSM Call) >>>>>>>>>> U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.16 8.1.200:5060;branch=z9hG4bK31bc676961f91e18..To: <sip:05353490056@192.168.1 .10>..From: <sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 8 76D1B912D91212@192.168.1.200..CSeq: 1 INVITE..Server: Sip EXpress router (0 .9.2 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.10:5060 "Nois y feedback tells: pid=4587 req_src_ip=192.168.1.200 req_src_port=5060 in_u ri=sip:05353490056@192.168.1.10 out_uri=sip:20105353490056@192.168.1.35:506 0 via_cnt==1".... # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.35:5060 INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route: <sip:192 .168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP 192.168.1.10;branch=z9h G4bK188d.03855495.0..Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK31bc 676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From: < sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D9121 2@192.168.1.200..CSeq: 1 INVITE..Contact: <sip:12345@192.168.1.200:5060>..S ession-Expires: 100;refresher=uac..Supported: timer..Content-Type: applicat ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-. .c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18 G72 9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 192.168.1.10:5060 -> 192.168.1.200:5060 SIP/2.0 200 ok -- no more pending branches..Via: SIP/2.0/UDP 192.168.1.200: 5060;branch=z9hG4bK31bc676961f91e18..To: <sip:05353490056@192.168.1.10>;tag =2f9bfc2acf470ceacf4efdebbaa289b4-026c..From: <sip:12345@192.168.1.10;user= phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D91212@192.168.1.200..CSeq: 1 CAN CEL..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length: 0..Wa rning: 392 192.168.1.10:5060 "Noisy feedback tells: pid=4586 req_src_ip=19 2.168.1.200 req_src_port=5060 in_uri=sip:05353490056@192.168.1.10 out_uri=s ip:20105353490056@192.168.1.35:5060 via_cnt==1"....

2 1

[Serusers] auth_db module table parameter?
by Cameron Beattie 03 Jun '05

03 Jun '05

I want SER to use another application's database but can't see a way to tell the auth_db module which table to use. I can see the paramaters for the host/database name, and the column names. But how do I tell SER which table to use or does it only ever use subscriber? Regards Cameron

2 2

[Serusers] CDR not being recorded at ACC table
by Felipe Martins 03 Jun '05

03 Jun '05

Hi everybody, I want to test my billing system based on the CDR data I've recorded at ACC table, but there is a problem, the data is not being recorded as I thought. I don't know why, when I go directly through mysql and make a "select * from acc;" it gives me: Empty set (0.00 sec) I don't know why the CDR date is not being recorded, I looked for it at the forum and google, and found something about mysql installation. It seems that I haven't compiled mysql with ACC support. Is that right ? Did someone have the same problem already ? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

2 1

[Serusers] avpops / failure route
by Martin Koenig 03 Jun '05

03 Jun '05

Hello, is it possible to use avp_write on the responses in failure route? If yes, Contact header field could be evaluated and pushed into RURI, then routing via route(x) to a new destination and finally solve the 3xx forwarding issue. Regards, Martin

2 3

[Serusers] ngrep of call transfer
by Iqbal 03 Jun '05

03 Jun '05

Hi Does anyone have a dump of what a succesful call transfer for the following two scenarios looks like PSTN <---> PHONE A PHONE A -----> PHONE B PSTN <----> PHONE B basically A transfer the call to B the next scenario is the same but instead of PSTN , its between IP only PHONE A <---> PHONE B PHONE B ---> PHONE C PHONE A <---> PHONE C tks Iqbal

1 0

[Serusers]
by Grigory Fishilevich 03 Jun '05

03 Jun '05

hi guys, is it possible to use a global phone book with ser? i found no features like this in sip rfcs, but this is imho very impotant thing for companys etc. do anybody know an way to implement a global phone book for all users of one ser? bg, grigory

4 7

[Serusers] redirecting SIP messages to SEMS
by Llanos Serna García-Conde 03 Jun '05

03 Jun '05

2 1

[Serusers] Pre-paid billing based upon SER
by Terje Myhre 03 Jun '05

03 Jun '05

Hello, We're an ip-telephony operator in Sweden (www.wx3.se) , and we're looking for a platform which can help us undertake pre-paid billing of IP-telephony users. We have ourselves a processing platform for CDRs, but we don't have a registrar/proxy (rtp-handling) / voice messaging (potentially asterisk) which can make us support pre-paid services. What we need is a platform which can make a request (including call details) to a billing system, and continue the call set-up if the response is positive. As the call is ongoing, the system will periodically continue to ask the billing system if the call can continue (for another 60 secs, for example). If the call cannot continue any longer, the call will be "taken down". If a caller call, and there is not enough money on the account, the billing system will return with a negative response and a sip-url, the caller should be directed to the sip-url - which belongs to a voice-mail system, in which the appropriate response will be played. So my question is: Can SER - possibly with support of RTP Router : to have control of RTP-flow / call-redirection - support these requirements. Br, Terje Myhre <http://www.wx3.se/> www.wx3.se sip: <mailto:tem@wx3.se> tem(a)wx3.se tel +46 8 5250 7972

2 1

[Serusers] ser don't start nedd help
by harry gaillac 03 Jun '05

03 Jun '05

Hello I've ever sent this question. What must i do in order to run ser. i tried ser_mysql.sh reinstall without success ! Regards harry serveur1:~# tail /var/log/syslog Jun 3 10:14:15 serveur1 ntpd[1912]: synchronized to 213.161.8.44, stratum 2 Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: Maxfwd module- initializing Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: permissions - initializing Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: WARNING: File not found: /etc/ser/ permissions.allow Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: Default allow file (/etc/ser/permi ssions.allow) not found => empty rule set Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: WARNING: File not found: /etc/ser/ permissions.deny Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: Default deny file (/etc/ser/permis sions.deny) not found => empty rule set Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: table_version(): Invalid number of rows received: 2, grp Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: group:mod_init(): Error while quer ying table version Jun 3 10:14:21 serveur1 /usr/sbin/ser[2565]: init_mod(): Error while initializi ng module group _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

2 3

[Serusers] starting ser
by harry gaillac 03 Jun '05

03 Jun '05

Hello, May be I'm tedious with my problems however i would like to get help from users . what's : "Invalid number of rows received" i installed: ser-mysql-module_0.10.99+cvs20050519_i386.deb ser_0.10.99+cvs20050519_i386.deb Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: Maxfwd module- initializing Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: permissions - initializing Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: WARNING: File not found: /etc/ser/permissions.allow Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: WARNING: File not found: /etc/ser/permissions.deny Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: table_version(): Invalid number of rows received: 2, trusted Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: permissions:init_trusted(): Error while querying table version Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: Error while initializing allow_trusted function Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: table_version(): Invalid number of rows received: 2, grp Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: group:mod_init(): Error while querying table version Jun 3 12:09:05 serveur1 /usr/sbin/ser[4952]: init_mod(): Error while initializing module group _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

2 1

Re: [Serusers] t_write_req
by Samuel Osorio Calvo 03 Jun '05

03 Jun '05

t_write_req appeared in SER version 0.9, in 0.8.12 it had another sintax, which I don't remember now...take a look at older posts or the tm module interface to find it out.. Samuel. Unclassified. >>> Llanos Serna García-Conde <llanosserna(a)hotmail.com> 06/03/05 12:44PM >>> Hi all, in the SER-SEMS-HOWTO, the next paragraph appears: Redirecting the messages is a very simple operation. You just have to call the t_write_unix() (or t_write_req()) function from the 'tm' module, which you have to load. In ser.cfg routing: t_write_req("/tmp/am_fifo", "voicemail") But in my tm module (SER 0.8.14), t_write_req doesn't appear, and so if I add that line I get an error of configuration. What's wrong with this? Please help, Llanos

1 0

Re: [Serusers] interconnect with ipcb
by Iqbal 03 Jun '05

03 Jun '05

Simply because it allows you then to use SER to terminate to them, and they then handoff the traffic, also you only need to open an account at one place, and billing is easier....Having said that I dont use them, I merely tested with them. Having said that it all depends on your setup, and exactly what it is you are trying to do, if you are happy with one endpoint, do it that way, tie up tightly with them, you see each gateway will have its own quirks, or have your own gateways, it depends on how flexibile you wish to be. Iqbal Ritesh Jalan wrote: >Hi Iqbal Sir > >Can you tell us how reliable are IPCB? as you have recomended them, further >they dont terminate calls, you need to buy mins from someone else. They only >have the hosted switch. then why we should use SER with IPCB?? > > >Thanks & Regards >Ritesh Jalan > > >----- Original Message ----- >From: "Iqbal" <iqbal(a)gigo.co.uk> >To: <serusers(a)lists.iptel.org> >Sent: Thursday, June 02, 2005 10:39 PM >Subject: [Serusers] interconnect with ipcb > > > > >>Hi >> >>A few people here were looking to terminate traffic using SER and >>looking for pstn end gateways, well if you sign up with ipcb.net (and no >>I am not connected to them..but hoping if they read this, i will get a >>pizza out of this :-)) you can connect SER to any of the gateways you >>wish worldwide. >> >>How they do it is simple, they have written a module which you compile >>into ser, load as normal, it has a user/pass in it, and u assign there >>IP address as trusted in ur config. Then when any requests are sent, the >>auth you using that user/pass, and let traffic flow. Works fine. >> >>Now I was wondering if some of the programmers on this list could take a >>look at the module (they will email it to you) surely it would be easy >>to do the same for any gateway out there....I guess >> >>Anyhow now all SER people can get their own gateways. >> >>Iqbal >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >. > > >

1 0

[Serusers] Re: ser don' start
by Greger V. Teigre 03 Jun '05

03 Jun '05

If you use the exact config file from onsip.org, you should have no problems. I have seen your other posts where people indicate that you have a problem with the DB, which is what these errors tell you: Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: bind_dbmod: Module sql does not export db_use_table function Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: ERROR: group_db_bind: unable to bind to the database module Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: init_mod(): Error while initializing module group You may have discrepencies between ser version and modules. Make sure you delete the full binary tree and do make proper before compiling new versions. This way you will know that the modules match the ser version. g-) harry gaillac wrote: > Hello > > I use onsip.org to write ser.cfg in order to use > asterisk as PSTN gateway. > > However i don't find docs for params permissions > module: > modparam("permissions", "db_mode", 1) => send errors > modparam("permissions", "trusted_table", "trusted") > > I try to load group module for ACL but ser don't start > > Need help > > Harry > > serveur1:/etc/ser# tail /var/log/syslog > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: Maxfwd > module- initializing > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: > permissions - initializing > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: WARNING: > File not found: /etc/ser/permissions.allow > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: Default > allow file (/etc/ser/permissions.allow) not found => > empty rule set > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: WARNING: > File not found: /etc/ser/permissions.deny > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: Default > deny file (/etc/ser/permissions.deny) not found => > empty rule set > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: > bind_dbmod: Module sql does not export db_use_table > function > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: ERROR: > group_db_bind: unable to bind to the database module > Jun 3 11:39:51 serveur1 /usr/sbin/ser[4299]: > init_mod(): Error while initializing module group > > > > > > > > _____________________________________________________________________________ > Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos > mails, photos et vidéos ! Créez votre Yahoo! Mail sur > http://fr.mail.yahoo.com

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[Serusers] Why don't response?
by Sandra Donaire 03 Jun '05

03 Jun '05

Hi!! I can`t run ok my server. It don't response the messages, and it running. I attach: * Config file * Capture of ethereal: "tramas.txt" * And logs when i run the server: "run.txt" If anybody can help me... Thanks! -- Sandra Donaire Arroyo Telefónica I+D División de "Tecnologías de acceso a redes IP" ading module /usr/local/lib/ser/modules/sl.so 0(6856) loading module /usr/local/lib/ser/modules/tm.so 0(6856) loading module /usr/local/lib/ser/modules/rr.so 0(6856) loading module /usr/local/lib/ser/modules/maxfwd.so 0(6856) loading module /usr/local/lib/ser/modules/usrloc.so 0(6856) loading module /usr/local/lib/ser/modules/registrar.so 0(6856) loading module /usr/local/lib/ser/modules/mysql.so 0(6856) loading module /usr/local/lib/ser/modules/auth.so 0(6856) loading module /usr/local/lib/ser/modules/auth_db.so 0(6856) loading module /usr/local/lib/ser/modules/pa.so 0(6856) loading module /usr/local/lib/ser/modules/msilo.so 0(6856) set_mod_param_regex: usrloc matches module usrloc 0(6856) set_mod_param_regex: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) set_mod_param_regex: usrloc matches module usrloc 0(6856) set_mod_param_regex: found <db_url> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) set_mod_param_regex: auth_db matches module auth_db 0(6856) set_mod_param_regex: found <calculate_ha1> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(6856) set_mod_param_regex: auth_db matches module auth_db 0(6856) set_mod_param_regex: found <password_column> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(6856) set_mod_param_regex: auth_db matches module auth_db 0(6856) set_mod_param_regex: found <db_url> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(6856) set_mod_param_regex: rr matches module rr 0(6856) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/local/lib/ser/modules/rr.so] 0(6856) set_mod_param_regex: pa matches module pa 0(6856) set_mod_param_regex: found <default_expires> in module pa [/usr/local/lib/ser/modules/pa.so] 0(6856) set_mod_param_regex: msilo matches module msilo 0(6856) set_mod_param_regex: found <db_url> in module msilo [/usr/local/lib/ser/modules/msilo.so] 0(6856) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(6856) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <record_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(6856) find_export: found <loose_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(6856) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <www_authorize> in module auth_db [/usr/local/lib/ser/modules/auth_db.so] 0(6856) find_export: found <www_challenge> in module auth [/usr/local/lib/ser/modules/auth.so] 0(6856) find_export: found <save> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(6856) find_export: found <m_dump> in module msilo [/usr/local/lib/ser/modules/msilo.so] 0(6856) find_export: found <lookup> in module registrar [/usr/local/lib/ser/modules/registrar.so] 0(6856) find_export: found <t_newtran> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <m_store> in module msilo [/usr/local/lib/ser/modules/msilo.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <sl_reply_error> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <m_store> in module msilo [/usr/local/lib/ser/modules/msilo.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) routing table 0: 0(6856) log( 0(6856) 4 0(6856) , "**Mensaje recibido****** " 0(6856) ); 0(6856) if ( 0(6856) NOT( 0(6856) external_module_call( 0(6856) f_ptr<0x28ce70> 0(6856) , "10" 0(6856) ); 0(6856) external_module_call( 0(6856) f_ptr<0............... " 0(6856) ); 0(6856) external_module_call( 0(6856) f_ptr<0x25c000> 0(6856) , "503" 0(6856) , "Service Unavailable" 0(6856) ); 0(6856) }; 0(6856) Listening on 10.95.24.118 [10.95.24.118]:5060 Aliases: anakin.hi.inet:5060 WARNING: no fork mode 0(6856) DEBUG: init_mod: sl_module stateless - initializing 0(6856) DEBUG: register_fifo_cmd: new command (sl_stats) registered 0(6856) DEBUG: MD5 calculated: 514c9da6346630a248fb02d697affea1 0(6856) DEBUG: init_mod: tm 0(6856) TM - initializing... 0(6856) Call-ID initialization: '7ebfb2b9' 0(6856) DEBUG: register_fifo_cmd: new command (t_uac_dlg) registered 0(6856) DEBUG: register_fifo_cmd: new command (t_uac_cancel) registered 0(6856) DEBUG: register_fifo_cmd: new command (t_hash) registered 0(6856) DEBUG: lock_initialize: lock initialization started 0(6856) DEBUG: register_fifo_cmd: new command (t_stats) registered 0(6856) DEBUG: MD5 calculated: 7a4a3882dbef725114aca71b5d0350e2 0(6856) DEBUG: MD5 calculated: bf040921a78c3fce97a0c4998426d78d 0(6856) DEBUG: init_mod: rr 0(6856) rr - initializing 0(6856) DEBUG: init_mod: maxfwd_module Maxfwd module- initializing 0(6856) DEBUG: init_mod: usrloc 0(6856) usrloc - initializing 0(6856) DEBUG: register_fifo_cmd: new command (ul_stats) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_rm) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_rm_contact) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_dump) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_flush) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_add) registered 0(6856) DEBUG: register_fifo_cmd: new command (ul_show_contact) registered 0(6856) find_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_free_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) mod_init(): Database connection opened successfuly 0(6856) DEBUG: init_mod: registrar 0(6856) registrar - initializing 0(6856) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) find_export: found <ul_bind_usrloc> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_register_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_get_all_ucontacts> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_insert_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_delete_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_get_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_lock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_unlock_udomain> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_release_urecord> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_insert_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_delete_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_get_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) find_export: found <ul_update_ucontact> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(6856) DEBUG: init_mod: mysql 0(6856) mysql - initializing 0(6856) DEBUG: init_mod: auth 0(6856) auth module - initializing 0(6856) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) DEBUG: init_mod: auth_db 0(6856) auth_db module - initializing 0(6856) find_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_free_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <pre_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(6856) find_export: found <post_auth> in module auth [/usr/local/lib/ser/modules/auth.so] 0(6856) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(6856) DEBUG: init_mod: pa 0(6856) Presence Agent - initializing 0(6856) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_newtran> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay_to_tcp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay_to_udp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_reply_with_body> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_is_local> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_get_trans_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_lookup_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_add_blind> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_forward_nonack> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request_within> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request_outside> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <new_dlg_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <dlg_response_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <new_dlg_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <dlg_request_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <free_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <print_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) DEBUG: init_mod: msilo 0(6856) MSILO: initializing ... 0(6856) find_export: found <db_use_table> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_init> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_close> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_raw_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_free_query> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_insert> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_delete> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <db_update> in module mysql [/usr/local/lib/ser/modules/mysql.so] 0(6856) find_export: found <load_tm> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <register_tmcb> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_newtran> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay_to_tcp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay_to_udp> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_reply> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_reply_with_body> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_is_local> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_get_trans_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_lookup_ident> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_add_blind> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_forward_nonack> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request_within> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request_outside> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <t_request> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <new_dlg_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <dlg_response_uac> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <new_dlg_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <dlg_request_uas> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <free_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(6856) find_export: found <print_dlg> in module tm [/usr/local/lib/ser/modules/tm.so] 0(0) fixing /usr/local/lib/ser/modules/maxfwd.so mf_process_maxfwd_header 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_send_reply 0(0) fixing /usr/local/lib/ser/modules/rr.so record_route 0(0) fixing /usr/local/lib/ser/modules/rr.so loose_route 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/auth_db.so www_authorize 0(0) fixing /usr/local/lib/ser/modules/auth.so www_challenge 0(0) fixing /usr/local/lib/ser/modules/registrar.so save 0(0) query="select username,contact,expires,q,callid,cseq,replicate,state,flags,user_agent from location " 0(0) preload_udomain(): Table is empty 0(0) fixing /usr/local/lib/ser/modules/msilo.so m_dump 0(0) fixing /usr/local/lib/ser/modules/registrar.so lookup 0(0) fixing /usr/local/lib/ser/modules/tm.so t_newtran 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/tm.so t_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/msilo.so m_store 0(0) fixing /usr/local/lib/ser/modules/tm.so t_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/tm.so t_reply 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/tm.so t_relay 0(0) fixing /usr/local/lib/ser/modules/sl.so sl_reply_error 0(0) fixing /usr/local/lib/ser/modules/msilo.so m_store 0(0) fixing /usr/local/lib/ser/modules/tm.so t_reply 0(0) fixing /usr/local/lib/ser/modules/tm.so t_reply 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) DEBUG: udp_init: trying SO_RCVBUF: 221184 0(0) DEBUG: setting SO_RCVBUF; set=221184,verify=221184 0(0) DEBUG: udp_init: trying SO_RCVBUF: 223232 0(0) DEBUG: setting SO_RCVBUF; set=223232,verify=221184 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 1(0) DEBUG: init_mod_child (-1): tm 1(0) DEBUG: callid: '7ebfb2b9-0(a)10.95.24.118' 1(0) DEBUG: init_mod_child (-1): usrloc 1(6858) DBG: open_uac_fifo: opening fifo... 1(6858) DEBUG: FIFO created @ /tmp/ser_fifo 1(6858) DEBUG: fifo /tmp/ser_fifo opened, mode=438 2(0) INFO: fifo process starting: 6860 2(0) DEBUG: init_mod_child (-2): tm 2(0) DEBUG: callid: '7ebfb2b9-0(a)10.95.24.118' 2(0) DEBUG: init_mod_child (-2): usrloc 1(6858) DEBUG: init_mod_child (-1): auth_db 2(6860) DEBUG: init_mod_child (-2): auth_db 1(6858) DEBUG: init_mod_child (-1): msilo 1(6858) MSILO: init_child #-1 / pid <6858> 2(6860) DEBUG: init_mod_child (-2): msilo 2(6860) MSILO: init_child #-2 / pid <6860> 1(6858) MSILO: child -1: Database connection opened successfully 2(6860) MSILO: child -2: Database connection opened successfully 0(6856) DEBUG: init_mod_child (1): tm 0(6856) DEBUG: callid: '7ebfb2b9-6856(a)10.95.24.118' 0(6856) DEBUG: init_mod_child (1): usrloc 2(6860) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 0(6856) DEBUG: init_mod_child (1): auth_db 0(6856) DEBUG: init_mod_child (1): msilo 0(6856) MSILO: init_child #1 / pid <6856> 0(6856) MSILO: child 1: Database connection opened successfully # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) #/* Uncomment these lines to enter debugging mode debug=8 fork=no log_stderror=yes #*/ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" fifo_mode=0666 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # Modulo para presencia loadmodule "/usr/local/lib/ser/modules/pa.so" # Modulo para el almacenamiento de mensajes loadmodule "/usr/local/lib/ser/modules/msilo.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- modparam("usrloc", "db_url", "sql://ser:sippv6@localhost/ser") # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # -- Parametros de "pa" -- modparam("pa", "default_expires", 3600) # Parametros para msilo modparam("msilo", "db_url", "sql://ser:sippv6@localhost/ser") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri=~portalv6.com) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("portalv6.com", "subscriber")) { www_challenge("portalv6.com", "0"); break; }; save("location"); #////////////////////////////// log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; #////////////////////////////////// break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { #sl_send_reply("404", "Not Found"); # ///////////////////////// if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; #/////////////////////////////////// break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #///////////////////////////////////// failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO: the downstream UA does not support MESSAGE requests ...\n"); # we have changed the R-URI with the contact address -- ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; }#////////////////////////////////////////////////////////////// me 5 (518 bytes on wire, 518 bytes captured) Arrival Time: Jun 2, 2005 08:39:15.000648000 Time delta from previous packet: 5.517287000 seconds Time since reference or first frame: 5.520523000 seconds Frame Number: 5 Packet Length: 518 bytes Capture Length: 518 bytes Ethernet II, Src: 00:01:03:18:66:50, Dst: 00:11:d8:8e:99:19 Destination: 00:11:d8:8e:99:19 (AsustekC_8e:99:19) Source: 00:01:03:18:66:50 (3com_18:66:50) Type: IP (0x0800) Internet Protocol, Src Addr: 10.95.13.103 (10.95.13.103), Dst Addr: 10.95.24.118 (10.95.24.118) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 0000 00.. = Differentiated Services Codepoint: Default (0x00) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 504 Identification: 0x9412 (37906) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 128 Protocol: UDP (0x11) Header checksum: 0x6a48 (correct) Source: 10.95.13.103 (10.95.13.103) Destination: 10.95.24.118 (10.95.24.118) User Datagram Protocol, Src Port: 1222 (1222), Dst Port: 5060 (5060) Source port: 1222 (1222) Destination port: 5060 (5060) Length: 484 Checksum: 0x5fd0 (correct) Session Initiation Protocol Request-Line: REGISTER sip:portalv6.com SIP/2.0 Method: REGISTER Resent Packet: False Message Header Via: SIP/2.0/UDP 10.95.13.103:7077 From: <sip:sandri@portalv6.com>;tag=ccff3170-7028-4b70-a4e1-ec34cf8515b5 SIP from address: sip:sandri@portalv6.com SIP tag: ccff3170-7028-4b70-a4e1-ec34cf8515b5 To: <sip:sandri@portalv6.com> SIP to address: sip:sandri@portalv6.com Call-ID: adc35f90-bb28-4c4f-8ec2-5f52e24c581f(a)10.95.13.103 CSeq: 1 REGISTER Contact: <sip:10.95.13.103:7077>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK" User-Agent: Windows RTC/1.0 Expires: 1200 Event: registration Allow-Events: presence Content-Length: 0 Frame 6 (415 bytes on wire, 415 bytes captured) Arrival Time: Jun 2, 2005 08:39:15.000854000 Time delta from previous packet: 0.000206000 seconds Time since reference or first frame: 5.520729000 seconds Frame Number: 6 Packet Length: 415 bytes Capture Length: 415 bytes Ethernet II, Src: 00:01:03:18:66:50, Dst: 00:11:d8:8e:99:19 Destination: 00:11:d8:8e:99:19 (AsustekC_8e:99:19) Source: 00:01:03:18:66:50 (3com_18:66:50) Type: IP (0x0800) Internet Protocol, Src Addr: 10.95.13.103 (10.95.13.103), Dst Addr: 10.95.24.118 (10.95.24.118) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 0000 00.. = Differentiated Services Codepoint: Default (0x00) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 401 Identification: 0x9413 (37907) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 128 Protocol: UDP (0x11) Header checksum: 0x6aae (correct) Source: 10.95.13.103 (10.95.13.103) Destination: 10.95.24.118 (10.95.24.118) User Datagram Protocol, Src Port: 1222 (1222), Dst Port: 5060 (5060) Source port: 1222 (1222) Destination port: 5060 (5060) Length: 381 Checksum: 0xc51d (correct) Session Initiation Protocol Request-Line: SUBSCRIBE sip:aragorn@portalv6.com SIP/2.0 Method: SUBSCRIBE Resent Packet: False Message Header Via: SIP/2.0/UDP 10.95.13.103:7077 From: "sandri" <sip:sandri@portalv6.com>;tag=cb57ec99-7a4a-493c-84f7-c77f3ef63602 SIP Display info: "sandri" SIP from address: sip:sandri@portalv6.com SIP tag: cb57ec99-7a4a-493c-84f7-c77f3ef63602 To: <sip:aragorn@portalv6.com> SIP to address: sip:aragorn@portalv6.com Call-ID: e8bc2af8-93d8-46e7-9e9c-72b56a6edc89(a)10.95.13.103 CSeq: 1 SUBSCRIBE Contact: <sip:10.95.13.103:7077> User-Agent: Windows RTC/1.0 Expires: 1800 Content-Length: 0 Frame 7 (409 bytes on wire, 409 bytes captured) Arrival Time: Jun 2, 2005 08:39:15.000856000 Time delta from previous packet: 0.000002000 seconds Time since reference or first frame: 5.520731000 seconds Frame Number: 7 Packet Length: 409 bytes Capture Length: 409 bytes Ethernet II, Src: 00:01:03:18:66:50, Dst: 00:11:d8:8e:99:19 Destination: 00:11:d8:8e:99:19 (AsustekC_8e:99:19) Source: 00:01:03:18:66:50 (3com_18:66:50) Type: IP (0x0800) Internet Protocol, Src Addr: 10.95.13.103 (10.95.13.103), Dst Addr: 10.95.24.118 (10.95.24.118) Version: 4 Header length: 20 bytes Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00) 0000 00.. = Differentiated Services Codepoint: Default (0x00) .... ..0. = ECN-Capable Transport (ECT): 0 .... ...0 = ECN-CE: 0 Total Length: 395 Identification: 0x9414 (37908) Flags: 0x00 0... = Reserved bit: Not set .0.. = Don't fragment: Not set ..0. = More fragments: Not set Fragment offset: 0 Time to live: 128 Protocol: UDP (0x11) Header checksum: 0x6ab3 (correct) Source: 10.95.13.103 (10.95.13.103) Destination: 10.95.24.118 (10.95.24.118) User Datagram Protocol, Src Port: 1222 (1222), Dst Port: 5060 (5060) Source port: 1222 (1222) Destination port: 5060 (5060) Length: 375 Checksum: 0x7380 (correct) Session Initiation Protocol Request-Line: SUBSCRIBE sip:saul@portalv6.com SIP/2.0 Method: SUBSCRIBE Resent Packet: False Message Header Via: SIP/2.0/UDP 10.95.13.103:7077 From: "sandri" <sip:sandri@portalv6.com>;tag=a42a2bea-5be8-400d-a52a-257beffc323f SIP Display info: "sandri" SIP from address: sip:sandri@portalv6.com SIP tag: a42a2bea-5be8-400d-a52a-257beffc323f To: <sip:saul@portalv6.com> SIP to address: sip:saul@portalv6.com Call-ID: 3e3717b5-4c34-4c9f-946b-59ef89508706(a)10.95.13.103 CSeq: 1 SUBSCRIBE Contact: <sip:10.95.13.103:7077> User-Agent: Windows RTC/1.0 Expires: 1800 Content-Length: 0

2 2

[Serusers] t_write_req
by Llanos Serna García-Conde 03 Jun '05

03 Jun '05

1 0

[Serusers] pb with jabber
by mma 03 Jun '05

03 Jun '05

i add the following entries on the sip_jab.jusers table insert into jusers values ('','user1jabber','password','sip:user1@sip.server.net',0); insert into jusers values ('','user2jabber','password','sip:user1@sip.server.net',0); i created user1jabber and user2jabber using exodus as jabber client i created also user1 and user2 on ser whiwc is running on 192.168.1.70 (sip.server.net) jabber is runninng also on the same machine : 192.168.1.70 (jabber.server.net) jabber works fine, i have tested it; DNS server : 195.167.203.168 i want the two users to communicate using IM user1 is connecting to windows messenger : user1(a)sip.server.net (192.168.1.118) user2 is connecting to exodus : user2jabber(a)jabber.server.net (192.168.1.127) user1 send an IM to user2jabber using user2jabber*jabber.org(a)jabber.server.net the problems are : -user2jabber don't receive what user1 sends. -also pa modules does not work : i have response 'bad request' from the server when a SUBSCRIBE request is sent (i added user3 as a contact to user1 ) please i need help regards ///////////////////////////////////////////////////////////////////////////////////// log using ethereal: No. Time Source Destination Protocol Info 25 12.150821 192.168.1.118 195.167.203.168 DNS Standard query A sip.server.net 26 12.153145 195.167.203.168 192.168.1.118 DNS Standard query response A 192.168.1.70 27 12.160197 192.168.1.118 192.168.1.70 SIP Request: REGISTER sip:sip.server.net 28 12.164436 192.168.1.70 192.168.1.118 SIP Status: 200 OK (1 bindings) 29 12.166088 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted (1 bindings) 30 12.169036 192.168.1.118 192.168.1.70 SIP Request: SUBSCRIBE sip:user3@sip.server.net 31 12.172514 192.168.1.70 192.168.1.118 SIP Status: 400 Bad Request ..... 171 21.841023 192.168.1.118 192.168.1.70 SIP/SDP Request: INVITE sip:user2jabber*jabber.org@jabber.server.net, with session description 172 21.843627 192.168.1.70 192.168.1.118 SIP Status: 202 Accepted 173 21.861278 192.168.1.118 192.168.1.70 SIP Request: ACK sip:user2jabber*jabber.org@jabber.server.net 174 21.862529 192.168.1.118 192.168.1.70 SIP Request: MESSAGE sip:user2abber*jabber.org@jabber.server.net (text/plain) 175 21.865357 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted ..... 297 22.446316 192.168.1.118 192.168.1.70 SIP Request: BYE sip:user2jabber*jabber.org@jabber.server.net ////////////////////////////////////////////////////////////////////////////////////// ser.cfg ***************** debug=4 # debug level (cmd line: -dddddddddd) #fork=yes # (cmd. line: -D) fork=no log_stderror=yes # (cmd line: -E) #log_stderror=no # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 listen=sip.server.net listen=192.168.1.70 fifo="/tmp/ser_fifo" #fifo="/tmp/ser_fifoql://ser:heslo@localhost/ser" # for more info: sip_router -h # ------------------ module loading ---------------------------------- #modules loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- # -- registrar params -- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) # -- usrloc params -- modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_mode", 0) # -- jabber params -- modparam("jabber","db_url","mysql://jabber:jabber@127.0.0.1/sip_jab") modparam("jabber","jaddress","jabber.server.net") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.server.net=*") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; if (uri=~"[@:]sip\.server\.net([;:].*)*") #if (uri=~"^sip:(.+@)?(192\.168\.1\.70|(jabber\.)?server\.net)([:;\?].*)?$") { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { log("XJAB:methode SUBSCRIBE \n"); if(jab_go_offline()) if (t_newtran()) { log("XJAB: subscribe registrar\n"); # subscribe("registrar"); handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; }; if (search("To:.*@jabber\.server\.net")) { log("XJAB : search for jabber.sofiphone.net /n"); if (! t_newtran()) { sl_reply_error(); break; }; if (method=="MESSAGE") { log("MESSAGE received -> manage it with XJAB\n"); if(search("\n:on")) { if (jab_go_online()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:off")) { if (jab_go_offline()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:join")) { if (jab_join_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if(search("\n:exit")) { if (jab_exit_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if (jab_send_message()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("503","Service Unavailable"); }; break; }; if (method=="SUBSCRIBE") { log("subscribe jabber /n"); # subscribe("jabber"); handle_subscription("jabber"); break; }; log("NON_Message request received for JABBER gateway->dropt!\n"); sl_send_reply("202","Accepted"); break; }; if (!t_relay()) { sl_reply_error(); }; #forward(uri:host,uri:port); } /////////////////////////////////////////////////////////////////////////////////////

1 0

Re: [Serusers] digest authentication without db
by Cesc Santasusana 03 Jun '05

03 Jun '05

Hi, We have been using dbtext for a while and it works fine. Only one problem: serctl does not support dbtext, only mysql. So, I added dbtext support into serctl a while ago ... sent the updated script to ser-dev (or serusers, don't remember). I think it never made it into the cvs or anything ... but maybe you want to dig it out of the mail list for you to use it ... This way you can add, remove, check subscribers, groups, alias, etc automatically. Regards, Cesc Unclassified >>> Francesco Fondelli <francesco.fondelli(a)gmail.com> 06/03/05 09:14AM >>> Thank you all for the tips, I had a look at dbtext and I think it almost fits my needs. I just have to change its 'subscriber' table format to match my file format. Ciao FF _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + ser-0.9.0-fc1 and it is running fine.
by fedora＠voice.dyndns.tv 03 Jun '05

03 Jun '05

rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + ser-0.9.0-fc1 and it is running fine. (ito.) ser-0.9.0-fc1.i386.rpm ser-0.9.0-fc1.src.rpm ser-jabber-0.9.0-fc1.i386.rpm ser-mysql-0.9.0-fc1.i386.rpm ser-radius-0.9.0-fc1.i386.rpm ///////////////////////////////////////////// http://www.voice.dyndns.tv sip:admin@iptel.org Saitama Japan Sip EXpress router Support Team voice(a)wmail.plala.or.jp /////////////////////////////////////////////

1 0

[Serusers] Auth Radius and draft-sterman-aaa-sip-00 question.
by Ricardo Martinez 03 Jun '05

03 Jun '05

Hello list. I have a question about the radius authentication message. As i can see in the radius authorize message reaching to my Radius Server the format of this message is as follow: Attributes: User-Name = "user1(a)mydomain.com" Digest-Attributes = "<10><10>user1" Digest-Attributes = "<1><23>mydomain.com" Digest-Attributes = "<2>*429f5a94dfac500699b5465aae863a390d5ebd92" Digest-Attributes = "<4>&sip:1234567@mydomain.com" Digest-Attributes = "<3><10>REGISTER" Digest-Attributes = "<5><6>auth" Digest-Attributes = "<9><10>00000037" Digest-Attributes = "<8><10>dbb06da4" Digest-Response = "08525b9e17e0ed25fccc61b104ff9e20" Service-Type = Sip-Session Sip-Uri-User = "1234567" NAS-IP-Address = 10.1.1.3 NAS-Port = 5060 my question is... the format of this radius auth message with the "Digest-Attributes" attributes is created by SER module auth_radius or by the radiusclient ? When this message reach to my server (by the way is a Radiator) i used a "patch" created by me to "understand" the differents Digest-Attributes and parse to the Digest-Uri, Digest-Method. Digest Real. etc that correspond, because my Radiator server seems not to parse the different Digest-Attributes. As i have seen in another post regarding to auth radius issues, the message in this format is understand perfect by for example the FreeRadius Server. Is there a special configuration (maybe with the dictionary file) to make this work?. What special feature do i need to set in the radius server? maybe the draft-sterman-aaa-sip-00 ?????? Is somone using Radiator with SER? I hope someone could help me. Thanks in advance Regards Ricardo Martinez.-

2 1

RE: [Serusers] Re: radiusclient
by Kamran Ahmad 03 Jun '05

03 Jun '05

any comments i have </usr/local/lib/ser/modules/auth_radius.so> but ser is giving me error Jun 2 13:11:13 SER ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: libradiusclient.so.2: cannot open shared object file: No such file or directory Jun 2 13:11:13 SER ser: parse error (26,13-14): failed to load module Jun 2 13:11:13 SER ser: set_mod_param_regex: No module matching auth_radius found | Jun 2 13:11:13 SER ser: parse error (49,18-19): Can't set module parameter Jun 2 13:11:13 SER ser: set_mod_param_regex: No module matching auth_radius found | Jun 2 13:11:13 SER ser: parse error (50,18-19): Can't set module parameter Jun 2 13:11:13 SER ser: parse error (192,37-38): unknown command, missing loadmodule? Jun 2 13:11:13 SER ser: parse error (223,37-38): unknown command, missing loadmodule? --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > I have used ser 0.8.12 for remote radius. > You have to give entry of RADIUSSERVER IP in > radiusclient.conf > And also you have give entry in server and client, a > shared secret, > testing123 with each others' IP. Radclient in SER > end does not required. > > I also tried to configure ser 0.8.14 with Remote > Radius, but faced some > problem. It was not sending the requests to remote > radius. I didn't worked > with that problem. Later if I get time, then I will > see it again. > > I am sending you the required > radiusclient-0.3.2.tar.gz for ser 0.8.12. > If you can configure ser 0.8.14, then please inform > me. > > shaikat > > > > > > -----Original Message----- > From: Kamran Ahmad [mailto:p_kami@yahoo.com] > Sent: Thursday, June 02, 2005 1:32 PM > To: Shaikat Mahmud; serusers(a)lists.iptel.org > Subject: RE: [Serusers] Re: radiusclient > > > > i want to install freeradius on 2nd mechine. > i have two mechines one for ser and 2nd mechine for > freeradius > > any comments > > Kamran Ahmad > > --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > > > Hi, > > > > For radclient exe, you have to install freeradius. > > It is not a part of radiusclient. > > > > shaikat > > > > > > -----Original Message----- > > From: Kamran Ahmad [mailto:p_kami@yahoo.com] > > Sent: Thursday, June 02, 2005 1:13 PM > > To: Shaikat Mahmud; serusers(a)lists.iptel.org > > Subject: RE: [Serusers] Re: radiusclient > > > > > > thanks for reply but my problem is that i dont > have > > radiusclient-0.3.2.tar.gz as recommended in this > > document. i tried radiusclient-0.3.3.tar, > > radiusclient-0.4.8.tar.gz. but after compiliation > > > > there is execuatable "radiusclient" but > > this document is saying > > > > #radclient -f digest localhost auth > <shared_secret> > > > > thats the problem > > > > thanks > > Kamran Ahmad > > > > --- Shaikat Mahmud <shaikat(a)stitel.com> wrote: > > > > > Hi, > > > > > > Please read the radius how to. > > > > > > http://www.iptel.org/ser/doc/ser_radius/ser_radius.html > > > It works fine. > > > > > > Shaikat > > > > > > -----Original Message----- > > > From: serusers-bounces(a)lists.iptel.org > > > [mailto:serusers-bounces@lists.iptel.org] On > > > Behalf Of Kamran Ahmad > > > Sent: Thursday, June 02, 2005 12:51 PM > > > To: serusers(a)lists.iptel.org > > > Subject: [Serusers] Re: radiusclient > > > > > > hello > > > > > > can any one tell me how to install radclient > > > > > > i tried radiusclient-0.3.3.tar , > > > radiusclient-0.4.8.tar.gz but nothing working > > > > > > > > > Kamran > > > > > > > > > > > > __________________________________ > > > Discover Yahoo! > > > Use Yahoo! to plan a weekend, have fun online > and > > > more. Check it out! > > > http://discover.yahoo.com/ > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > __________________________________ > > Discover Yahoo! > > Find restaurants, movies, travel and more fun for > > the weekend. Check it out! > > > > http://discover.yahoo.com/weekend.html > > > > > > > > > > > __________________________________ > Discover Yahoo! > Get on-the-go sports scores, stock quotes, news and > more. Check it out! > http://discover.yahoo.com/mobile.html > __________________________________ Discover Yahoo! Use Yahoo! to plan a weekend, have fun online and more. Check it out! http://discover.yahoo.com/

2 1

[Serusers] digest authentication without db
by Francesco Fondelli 03 Jun '05

03 Jun '05

Hello, I'm new to SER and SIP as well so please forgive my mistakes. I'm trying to setup SER in order to forward calls to a pool of pstn gateways. I want use digest authentication for UAs but I cannot store userid and passwords on a db. Basically I would like to do: if (!www_authorize("mydomain.com", "subscriber")) { www_challenge("mydomain.com", "0"); break; }; getting userid and password from a text configuration file which contains such infos. How can I do that? I have written a ser cfg file and I would like someone tell me if is ok. Is a mix of several different cfg files I have found on the net. I'm sure is far to be ok :-) Thank for your help. Ciao ------------------------------------------------------------------- # ----------- global configuration parameters ------------------------ #debug=3 debug=4 #fork=yes fork=no #log_stderror=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #uid= #gid= listen=192.168.1.114 # alias="mydomain.com" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc", "db_mode", 0) # ------------------------- routing logic --------------------------- route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { log("Too many hops\n"); sl_send_reply("483","Too Many Hops"); break; }; if ( msg:len > max_len ) { log("Message too big\n"); sl_send_reply("513", "Message too big"); break; }; # process requests for our domain (gws included) if (uri=~"[@:]mydomain\.com([;:].*)*" | uri=~"@192.168.1.171([;:].*)*" | #pstn gw1 uri=~"@192.168.1.172([;:].*)*" | #pstn gw2 uri=~"@192.168.1.173([;:].*)*" ) { #pstn gw3 log("Request is for mydomain.com\n"); # registers always MUST be authenticated to # avoid stealing incoming calls if (method=="REGISTER") { log("Request is REGISTER\n"); if (!www_authorize("mydomain.com", "subscriber")) { log("REGISTER has no credentials, sending challenge\n"); www_challenge("mydomain.com", "0"); break; }; # prohibit attempts to grab someone else's address # using someone else's valid credentials if (!check_to()) { log("Cheating attempt\n"); sl_send_reply("401", "Unauthorized"); break; }; # update user location database (it should be in mem) log("REGISTER is authorized, saving location\n"); save("location"); break; }; # now it's about PSTN destinations through our gateways if (uri=~"sip:[0-9]+@.*") { # all PSTN destinations only for authenticated users # (GWs, which have no digest support, are authenticated # by its IP address) if (!(src_ip==192.168.1.171 | #pstn gw1 src_ip==192.168.1.172 | #pstn gw2 src_ip==192.168.1.173) & #pstn gw3 !(www_authorize("mydomain.com", "subscriber"))) { www_challenge("mydomain.com", "0"); break; }; # requests to gateways must be record-route because the GWs accept # only requests coming from our proxy if (method=="INVITE") record_route(); # XXX: find the best gw using first part of telephone number and... rewritehostport("192.168.1.171:5060"); #172 or 173 } else { # native SIP destinations are handled using our USRLOC DB # and are allowed only from gws if (src_ip==192.168.1.171 | #pstn gw1 src_ip==192.168.1.172 | #pstn gw2 src_ip==192.168.1.173) { #pstn gw3 if (!lookup("location")) { log("Unable to lookup contact, sending 404\n"); sl_send_reply("404", "Not Found"); break; }; } else { log("No native SIP destination allowed\n"); sl_send_reply("403", "Permission denied"); break; }; }; } else { # outbound requests are not allowed log("No outbound requests allowed\n"); sl_send_reply("403", "Permission denied"); break; }; # and finally.. forward to current uri; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if(!t_relay()) { sl_reply_error(); }; }

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[Serusers] SER 0.9 | How to check URI against uri table ?
by lamasse＠free.fr 03 Jun '05

03 Jun '05

Hi SER Users ! Is there a mean to check that the URI submitted in 'TO' header during registration is allowed for a user in the uri table or SER(if URI and username are not the same) ?? I tried with function check_to() but it doesn't work. REGISTERs are rejected because username does not match with URI. However it's written in documentation that this function should performs a check in the uri table. Is there any other me to perform this verification ? Thanks for your help !

2 1

[Serusers] SER connect to Clarent C5CM
by root linux 03 Jun '05

03 Jun '05

Hi, Does anybody have successful connect SER to Clarent C5CM? Regards, rootlinux __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] 0.9.0 Invalid table version
by harry gaillac 02 Jun '05

02 Jun '05

Hello serusers, I looked this mail : http://lists.iptel.org/pipermail/serusers/2005-March/016424.html However it don't help me. I download : ser-mysql-module_0.10.99+cvs20050519_i386.deb ser_0.10.99+cvs20050519_i386.deb What can i do ? change version table? Regards Harry serveur1:~# tail /var/log/syslog Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: WARNING: File not found: /etc/ser/permissions.deny Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: table_version(): Invalid number of rows received: 2, trusted Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: permissions:init_trusted(): Error while querying table version Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: Error while initializing allow_trusted function Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: table_version(): Invalid number of rows received: 2, grp Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: group:mod_init(): Error while querying table version Jun 2 23:11:46 serveur1 /usr/sbin/ser[2624]: init_mod(): Error while initializing module group Jun 2 23:17:01 serveur1 /USR/SBIN/CRON[2725]: (root) CMD ( run-parts --report /etc/cron.hourly) _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

2 1

[Serusers] ser don't start
by harry gaillac 02 Jun '05

02 Jun '05

hello, what's wrong with module permissions and group ? Harry serveur1:/tmp# tail /var/log/daemon.log Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: Maxfwd module- initializing Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: permissions - initializing Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: table_version(): Invalid number of rows received: 2, trusted Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: permissions:init_trusted(): Error while querying table version Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: Error while initializing allow_trusted function Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: table_version(): Invalid number of rows received: 2, grp Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: group:mod_init(): Error while querying table version Jun 2 21:13:52 serveur1 /usr/sbin/ser[5058]: init_mod(): Error while initializing module group _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

3 3

[Serusers] Hiring programmers
by Rosa De Santis 02 Jun '05

02 Jun '05

Hello all. We need to hire programmers to help us implement a SER + SEMS server, java based Sip Softphone and some other related programs. We are a US based VoIP company. Our website is www.vivophone.com Please, contact us for more information Thanks in advance Rosa

1 0

[Serusers] Tr: problem with auth_db module
by harry gaillac 02 Jun '05

02 Jun '05

Remarque : message transféré en pièce jointe. _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] problem with auth_db module
by harry gaillac 02 Jun '05

02 Jun '05

Hello, I have a problem with auth_db module when ser start. However ser start (ser.cfg should be ok) but ser_fifo is missing. Need Help Harry serveur1:/etc/ser# /etc/init.d/ser restart Restarting ser: serListening on udp: 80.119.7.156 [80.119.7.156]:5060 tcp: 80.119.7.156 [80.119.7.156]:5060 Aliases: tcp: 156.7.119-80.rev.gaoland.net:5060 udp: 156.7.119-80.rev.gaoland.net:5060 *: nxs.yi.org:* . serveur1:/etc/ser# tail /var/log/syslog Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Maxfwd module- initializing Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: permissions - initializing Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) Jun 2 19:25:18 serveur1 /usr/sbin/ser[3053]: ERROR: fix_expr : fix_actions error _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] interconnect with ipcb
by Iqbal 02 Jun '05

02 Jun '05

Hi A few people here were looking to terminate traffic using SER and looking for pstn end gateways, well if you sign up with ipcb.net (and no I am not connected to them..but hoping if they read this, i will get a pizza out of this :-)) you can connect SER to any of the gateways you wish worldwide. How they do it is simple, they have written a module which you compile into ser, load as normal, it has a user/pass in it, and u assign there IP address as trusted in ur config. Then when any requests are sent, the auth you using that user/pass, and let traffic flow. Works fine. Now I was wondering if some of the programmers on this list could take a look at the module (they will email it to you) surely it would be easy to do the same for any gateway out there....I guess Anyhow now all SER people can get their own gateways. Iqbal

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