- sr-users - kamailio.org

[Users] no voice when using nathelper with rtpproxy
by messa＠innsof.com 16 Nov '05

16 Nov '05

Hi all I install rtpproxy in the same host (with the public IP) than openser. the installation of rtpproxy put rtpproxy.sock in /var/run directory.But when i make a call between 2 clients (each client in a different LAN), i have no voice after signalisation. And openser give me these errors: [root@Is-dev01 openser-1.0.0-tls]# /usr/local/sbin/openser Listening on udp: 62.128.173.101 [62.128.173.101]:5060 tcp: 62.128.173.101 [62.128.173.101]:5060 Aliases: tcp: 62-128-173-101.iwayafrica.com:5060 udp: 62-128-173-101.iwayafrica.com:5060 WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 2(0) INFO: fifo process starting: 20621 2(0) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(0) INFO: no fifo_db_url given - fifo DB commands disabled! 0(20619) ERROR: force_rtp_proxy2: no available proxies 0(20619) ERROR: force_rtp_proxy2: no available proxies 0(20619) ERROR: unforce_rtp_proxy: no available proxies What's the matter? Best regards Serge

3 2

[Serusers] Mediaproxy High Load
by tcchan 16 Nov '05

16 Nov '05

Hi All, I tested the load of the mediaproxy (ver 1.4.2) and found that it is quite high. To give an idea, on a P4 2.4GHz machine, 20 sessions give a reading of about 35%, and it will go to almost 90% for 50 sessions. If we have a system that say can accommodate 10000 users (behind nat) or concurrent calls, that probably means we need about 10000/50=200 machines. That is way too many boxes. Does anybody has a solution for this? Thanks for any help available. Regards, TC Chan

1 0

[Serusers] Incoming message
by Lazy.charles 16 Nov '05

16 Nov '05

1 0

[Serusers] Inaccurate Radius Accounting
by lenirsantiago＠yahoo.com 16 Nov '05

16 Nov '05

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir

1 0

[Users] Inaccurate Radius Accounting
by lenirsantiago＠yahoo.com 16 Nov '05

16 Nov '05

Hey guys, I'm using Radius to do pretty much everything regarding my users, i.e. registrations, authorization, features and preferences. I implemented my own way of doing speeddial by using AVPs provided by radius and AVPOPs module, Here is the snippet: AVP-SIP = speedial:30:15615551212 avp_subst("s:caller_speedial/i:99", "/(.*):(.*)/\2/"); #15615551212 avp_printf("i:98","$rU:*"); #30 avp_printf("i:97","sip:$avp($temp2)@$td"); # sip:15615551212@xx.xx.xx.xx avp_printf("i:96","<sip:$avp($temp2)@$td>\r\n"); # <sip:15615551212@xx.xx.xx.xx>. xlog("L_N"," SPEEDIALING: $fU is speed-dialing $hdr(To)->$avp($temp2)\n"); if (avp_check("s:caller_speedial", "fm/$temp1/g")) { #avp_pushto("$To","i:96"); #avp_pushto("$ruri","i:97"); #avp_pushto("$duri","i:97"); #subst('/^To: (.*)(a)(.*)/To: "$avp($temp2)" <$avp($temp2)@\2/'); #append_hf("To: \"5615551212\" <15615551212(a)xx.xx.xx.xx>;user=phone"); #subst_uri('/^sip:(.*)@(.*)$/sip:$avp($temp2)@\2/i'); #subst_user('/(.*)/$avp($temp2)/'); #remove_hf("To"); rewriteuser("15615551212"); xlog("L_N"," SPEEDIALING: $fU is dialing $hdr(To) $avp($temp2)\n"); avp_delete("$temp1/g"); avp_delete("$temp2/g"); }; In this scenario, im forcing it instead of getting the value from AVPs just for ease of showing my point. The problem that I'm having is that no matter what I do to change the To header or change the R-URI, radius ALWAYS sends the accounting records with the original R-URI, in my case 30(a)xx.xx.xx.xx, how can I get acc.so to send the final resulting URI (15615551212(a)xx.xx.xx.xx) in the Called-Station-Id AVP? rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=97, length=298 Acct-Status-Type = Start Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = Invite User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway;user=phone" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "102" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 rlm_sql (sql): Reserving sql socket id: 12 rlm_sql (sql): Released sql socket id: 12 Sending Accounting-Response of id 97 to xx.xx.xx.xx:39216 rad_recv: Accounting-Request packet from host xx.xx.xx.xx:39216, id=98, length=292 Acct-Status-Type = Stop Service-Type = Sip-Session Acct-Terminate-Cause = 200 Error-Cause = 8 User-Name = "1000(a)xx.xx.xx.xx" Calling-Station-Id = "sip:1000@xx.xx.xx.xx" Called-Station-Id = "sip:30@xx.xx.xx.xx;user=phone" Sip-Translated-Request-URI = "sip:949449852150#15615551212@gateway:5060" Acct-Session-Id = "000d2890-d47f01e8-1dcc30bc-7a49c7a4(a)yy.yy.yy.yy" Sip-To-Tag = "230383F4-2AF" Sip-From-Tag = "000d2890d47f20f54576a906-290bdc33" Sip-CSeq = "103" NAS-IP-Address = xx.xx.xx.xx NAS-Port = 5060 Acct-Delay-Time = 0 Thanks in advance! Lenir

1 0

RE: [Serusers] LCR question
by Bjorn Asmul 16 Nov '05

16 Nov '05

What indexes would you recommend on the LCR table? -- Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Juha Heinanen Sent: Saturday, November 12, 2005 8:48 AM To: Stefan Prelle Cc: serusers(a)lists.iptel.org Subject: [Serusers] LCR question Stefan Prelle writes: > I am having huge performance problems. The OpenSER drops to effectivly > 1 CPS. My lcr table contains only 20 entries and I have just 2 gateways > configured, so this shouldn't be a problem. currently load_gws() makes a complex mysql query, but only one mysql query per second seems very slow to me. load_gws could be rewritten totally in c, but so far nobody has had time to do so. -- juha _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] Am I looking in the wrong place for SEMS?
by sip 15 Nov '05

15 Nov '05

I thought I'd regrab code for SEMS lately and make sure I had the most up to date source, but I noticed the sems.berlios.de page is conspicuously blank. Has the page for SEMS been moved somewhere? While I'm at it, does anyone know of a conference-capable application (hacked SEMS, Asterisk, other) that does multicast RTP? Just playing around with a few ideas, so I was curious if anyone's run across one or two. N.

1 0

[Serusers] Usrloc question
by Darren Nay 15 Nov '05

15 Nov '05

Hey all, I have a question regarding usrloc. I have run into a problem.. We have static routes sent to an asterisk server for all of our SIP usernames. In addition our IAD's will register with the same username, so that calls coming into our switch for that username will be routed to both the asterisk box and the SIP IAD. This way, whichever endpoint (IAD or asterisk) answers the call first will take the call. For example. root:/ # serctl ul show +18646404810 <sip:+18646404810@192.168.1.60>;q=1;expires=-1012151 <sip:+18646404810@192.168.1.157:5060>;q=;expires=403 192.168.1.60 is the asterisk server. This is a static route added by serctl. 192.168.1.157 is my IAD which registers with the switch every 10 minutes. So when calls are made to (864) 640-4810 then SER will send an INVITE to both location. I explained all of this just to explain now what my problem is, and ask if anyone may know a possible solution. Now, we also use asterisk to perform call fwd'ing functions. Asterisk will answer the call and then originate another call out back to SER to a new location. Now the problem! (finally!) This call fwd'ing method works very well in most cases, except that if the call fwd'ing is being sent to another location registered with SER then it will be redirected back to asterisk again, albeit to a different URI, and asterisk will kill the call because it thinks that it has looped (which I guess it has. sort of). So, I'm wondering if there is possibly a way to retrieve only the usrloc locations that don't contain the IP address 192.168.1.60 in the contact URI? This way I can just check if the src_ip is 192.168.1.60 and if so then retrieve all the usrloc locations - without asterisk - and the call will not be redirected back to asterisk. Is this possible? Or if anyone has any other ideas that may help then I am definitely open to suggestions. Thanks for your help!! Darren Nay Ionosphere, Inc VoIP Network Development dnay(a)ionosphere.net

2 1

[Serusers] Info on t_fork();
by Leo Papadopoulos 15 Nov '05

15 Nov '05

Dear Serusers: I cannot seem to find information on t_fork. There is only a small reference to it the the TM.so documentation, but no description of t_fork itself. Does anyone know where I can find information about this? Leo P.

1 0

RE: [Serusers] SEMS:announcement plug in
by rajesh.rajguru＠wipro.com 15 Nov '05

15 Nov '05

Hi, We are trying to forward INVITE to SEMS and play announcement. It doesn't play anything. The UA which is sending INVITE is getting time out But the same .wav file for announcement if I open from sound recorder manually ,then it plays. How to check that there is problem with codec ? Regards, Rajesh -----Original Message----- From: Xavier TRENTIN [mailto:xtrentin@softiphone.net] Sent: Monday, November 14, 2005 10:00 PM To: Rajesh Rajguru (WT01 - Voice & Next Generation Networks) Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SEMS:announcement plug in Does it play nothing or does it hang off? Perhaps the problem could be the Codec you used to record the file. B.R. Xavier. rajesh.rajguru(a)wipro.com a écrit : >*********************** >Pas de virus detecté dans l'attachement no filename > >Votre Mail a été vérifié par InterScan. >***********-*********** > > > >Hi, > >In our ser.cfg following is mentioned for announcement > >---- > if(!t_write_req("/tmp/am_fifo","announcement")) { > t_reply("500","error contacting sems"); >----- > >we are using fifo for communicating between ser and sems. >ser and sems are running on the same server (Fedora) When we start >sems, the default announcement file mentioned in sems.conf >(default_announce=default_en.wav) is not played. >Also there is no error in /var/log/messages wrt announcement plug in > >What could be the problem ? > >How does sems decides which sound recorder to play ? > > >The ll /tmp/am_fifo shows following > > >prw-r--r-- 1 root root 0 Nov 14 09:48 /tmp/am_fifo > >Regards, >Rajesh > > > > >Confidentiality Notice > > >The information contained in this electronic message and any >attachments to this message are intended for the exclusive use of the >addressee(s) and may contain confidential or privileged information. If >you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments. > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > > Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Users] a rtpproxy!
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi effectively, the first error is " ca't connect to rtpproxy". Where can i download a rtpproxy? Can i installed rtpproxy on the same host? regards! Serge!

2 1

[Serusers] [Help] Setting up SER in Redirect mode
by anonymous anonymous 15 Nov '05

15 Nov '05

Hello all, I am a student currently studying SIP with SER on a Slackware 10.2 system. I could have my test network working fine in proxy/registrar mode but I am having trouble in configuring SER for redirect mode. Actually I am totally new at SER administration and don't feel really comfortable in implementing code blocks into ser.cfg When I use the following ser.cfg file, SER simply ignore the redirect block and call alice at 192.168.1.2 whereas the call should be redirected to 192.168.1.4 I've been driftng all around the Internet gathering hints and tips for a few days but I'm still at that point. I guess it's a programming problem but can't find what's wrong exactly. Thank you in advance for your help. Matthieu Here is my uname -a result: Linux PC1 2.4.31 #6 Sun Jun 5 19:04:47 PDT 2005 i686 unknown unknown GNU/Linux My SER build: version: ser 0.9.4 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 15:54:20 Sep 6 2005 with gcc 4.0.1 My ser.cfg file based upon default ser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=8 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ listen=192.168.1.1 check_via=yes # (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 >= 2048 ) { 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==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; }; # redirect block, maybe at the wrong place if (uri=="sip:alice@192.168.1.2") { rewritehost("192.168.1.4"); sl_send_reply("302", "Redirect"); } }; 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(); }; } _________________________________________________________________ Recherchez sur votre PC comme sur le Web avec Windows Desktop Search ! http://desktop.msn.fr/

3 2

[Serusers] Concurrent call limitation
by Marcello Lupo 15 Nov '05

15 Nov '05

Hi to all, any idea as to limit the concurrent calls outgoing to a particular PSTN gateway supposing to let ser becoming dialogs aware so to have the status of active open calls?? Thanks, Bye, Marcello

1 0

[Users] Usrloc question
by Darren Nay 15 Nov '05

15 Nov '05

Hey all, I have a question regarding usrloc. I have run into a problem.. We have static routes sent to an asterisk server for all of our SIP usernames. In addition our IAD's will register with the same username, so that calls coming into our switch for that username will be routed to both the asterisk box and the SIP IAD. This way, whichever endpoint (IAD or asterisk) answers the call first will take the call. For example. root:/ # serctl ul show +18646404810 <sip:+18646404810@192.168.1.60>;q=1;expires=-1012151 <sip:+18646404810@192.168.1.157:5060>;q=;expires=403 192.168.1.60 is the asterisk server. This is a static route added by serctl. 192.168.1.157 is my IAD which registers with the switch every 10 minutes. So when calls are made to (864) 640-4810 then SER will send an INVITE to both location. I explained all of this just to explain now what my problem is, and ask if anyone may know a possible solution. Now, we also use asterisk to perform call fwd'ing functions. Asterisk will answer the call and then originate another call out back to SER to a new location. Now the problem! (finally!) This call fwd'ing method works very well in most cases, except that if the call fwd'ing is being sent to another location registered with SER then it will be redirected back to asterisk again, albeit to a different URI, and asterisk will kill the call because it thinks that it has looped (which I guess it has. sort of). So, I'm wondering if there is possibly a way to retrieve only the usrloc locations that don't contain the IP address 192.168.1.60 in the contact URI? This way I can just check if the src_ip is 192.168.1.60 and if so then retrieve all the usrloc locations - without asterisk - and the call will not be redirected back to asterisk. Is this possible? Or if anyone has any other ideas that may help then I am definitely open to suggestions. Thanks for your help!! Darren Nay Ionosphere, Inc VoIP Network Development dnay(a)ionosphere.net

1 0

[Serusers] Illegal redirection
by Joao Pereira 15 Nov '05

15 Nov '05

Hello I have two Cisco 7940 phones with private addresses (10.0.11.239 and 10.0.11.140) connected to SER also with private address (10.0.0.135), but in another network. My SER is with the default configuration. Between the networks I have a Checkpoint Firewall-1NG The Cisco IP phones can register because the REGISTER packets arent blocked. But the INVITEs never reach SER (I checked with ngrep), because the Firewall drops them, saying there was an illegal redirection. The most strange part, is that, when I try to make a phone call from PhoneA(10.0.11.239) to PhoneB(10.0.11.240), the INVITE is dropped before reaching SER, and it says "Illegal redirection 10.0.0.135->10.0.11.240". How can the firewall know that the INVITE was going to be redirected by SER to PhoneB(10.0.11.240) ???? my ser.cfg (the default one): # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 andrei Exp $ # simple quick-start config script # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen = 10.0.0.135 /* 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) # ------------------ 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/textops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 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") { 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(); }; }

3 4

[Users] OpenSER Clients
by Gary Major 15 Nov '05

15 Nov '05

2 1

[Serusers] dynamic sip port for clients behind same firewall
by Imran Akbar 15 Nov '05

15 Nov '05

Hi, The problem I'm facing is that if I have multiple clients behind the same firewall, calls between them don't work as each is using the same sip port. The clients can be made to use a random port, but the registrar server has to listen on all ports and respond to the client using the port it was sent on... does SER support this? Is there any other way to solve this problem? Thanks, Imran

2 1

Re: [Users] Forwarding..
by Bogdan-Andrei Iancu 15 Nov '05

15 Nov '05

Hi Sebastian, not sure if you can do this - a kind of middle registrar; but you can try to experiment by using the latest UAC module which provides the FROM mangling and authentication features. http://www.openser.org/docs/modules/1.1.x/uac.html regards, bogdan BASTARD Sebastien wrote: > Ok, it seems to work...but I have another question: > I would like to have a static configuration on the PC on a LAN with a > local account test1(a)test.net <mailto:test1@test.net>. There is openser > running on a computer of this LAN which is managing domain test.net > <http://test.net>. > I would like to have an association of accounts on the openser server > (I know asterisk is able to do it): > SIP client connect to my local openser with account test1(a)test.net > <mailto:test1@test.net> > OPENSER associates test1(a)test.net <mailto:test1@test.net> with > test1(a)iptel.org <mailto:test1@iptel.org> and inform iptel.org > <http://iptel.org> that this account is managed locally: it's like > openser act as a client / iptel ser server. > Then I can be called from outside without problem. > > Do you think it's possible to do it or to implement a module for that. > > Thanks a lot for your help > Sebastien > > On 11/9/05, *BASTARD Sebastien* <sebastien.bastard(a)gmail.com > <mailto:sebastien.bastard@gmail.com>> wrote: > > Ok, i'm going to try. > > Thanks for your help. > > sebastien > > > On 11/8/05, *Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > if I'm not wrong, you are talking about routing outbound requests > (requests which are not targeting your domain). If so, the > routing will > be done via DNS based on the domain name; you do not have to > specify > each of them. Just do like: > > if (!uri==myself) { > # this is an outbound call > t_relay(); > exit; > } > # process inbound calls > ........ > > regards, > bogdan > > BASTARD Sebastien wrote: > >> Hi, >> >> I would like to make run openser as a sip client when a client is >> connecting to it but with an unmanaged domain. I tried with the >> following configuration but with no result: >> >> if (!uri==myself) { >> # mark routing logic in request >> append_hf("P-hint: outbound\r\n"); >> # if you have some interdomain connections via TLS >> if(uri=~" iptel.org <http://iptel.org>") { >> forward("iptel.org <http://iptel.org>",5060); >> exit; >> } #else if(uri=~"@tls_domain2.net") { >> # t_relay_to_tls("IP_domain2","port_domain2"); >> # exit; >> #} >> route(1); >> }; >> >> Thanks for any help. >> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org <mailto:Users@openser.org> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

[Users] Exec module: Use output of an external command within SER?
by Joachim Fabini 15 Nov '05

15 Nov '05

Hi, A newbie question: Is there a way for OpenSER to use SIP headers (other than the URI set) that have been modified/processed by an external command (exec_dst, exec_msg)? Reading the documentation and mailing list archives revealed some questions similar to mine but no answer. The problem is: Delegate the change of one or more SIP message headers (or the whole SIP message) to an external application via exec module. Two straight-forward solutions: 1. exec_rmsg() - for "replace message". A merge of the functionality of exec_dst() and exec_msg(), i.e. OpenSER uses the output of exec_rmsg(command) to replace the current SIP message. This is imho an extremly flexible solution but slightly difficult to implement as it requires at least parsing the command's output - similar to a message received by OpenSER. 2. The command executed by exec_msg() can modify the environment variables SIP_HF_<hf_name> and OpenSER reuses these values after the exec() returned control to SER. Setvars could be used to switch between the current, read-only meaning of env-variables (e.g., modparam("exec", "setvars", 1)) and a read-write behavior (e.g. modparam("exec", "setvars", 2). This solution does not require any special handling except that OpenSER must read back the content of all environment variables which were passed to the command. Probably quite easy to implement but has the drawback that no new header fields can be added. Imho no major limitation as users can add their (dummy) header fields prior to invoking the command via exec() Or is there any other existing/planned procedure for SER to use the output of external commands for rewriting an existing SIP message. tia best regards --Joachim

5 9

[Users] Does openser support fax ?
by Nguyen Duc Phi 15 Nov '05

15 Nov '05

Hi all, Does openser support fax? if it support How can I config openser ? Best Regards.

4 4

[Users] t_relay to multiple path
by Jayesh Nambiar 15 Nov '05

15 Nov '05

hi everyone, In my configuration, I trust some source ips and if call comes from these src ips, without any authentication they are t_relayed to a gateway ip address for pstn termination. Is there any potential problems by trusting some clients by src_ips, I do so as I dont want them to login from anywhere else. I also wanted to know that if I do not get reply from a particular gateway, can I forward them to another gateway. I dont want to use LCR for this. I thought of using failure route, but the call does not fail it keeps on sending invite. What I wanted to do is, if I dont get a reply within 5 seconds, it will try the next t_relay_to_udp statement. the logic goes something like this: if(allow_trusted()) { if(uri=="sip:011...") { t_relay_to_udp("to a pstn gateway"); # if no reply for 5 seconds t_relay_to_udp("to another gateway) break; }; }; can somebody pls help on this. I am currently using 0.9.5 and will soon be upgrading to 1.0.0. Thanks a lot in advance Jayesh --------------------------------- Enjoy this Diwali with Y! India Click here

3 2

[Users] what about mediaproxy module?
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I want to know how i can configure the NAT helper module to make call between 2 LANs separated by internet,or simply to do Nat Transversal? Can the mediaproxy module help me to do NAT transversal? Regards!

2 1

[Users] how to configure openser to use nathelp
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I install openser in a host with private ip. i can make a call between clients in the same LAN. But now, i install openser in a host with public IP. I want to make a call between a client in LAN A and a client in a LAN B. the 2 LANs are separated by Internet.What can i use the nathelp module and configure openser to make this call possible? help me please!

2 1

[Users] Multi-domain setup problem
by Daryl Sanders 15 Nov '05

15 Nov '05

Hi Everyone, I just added a second domain to my openser box and screwed things up somehow. Maybe someone can help figure out what's going on. I'm using the new onsip config (unmodified except IPs & replacing breaks w/returns). I'm using mediaproxy for NAT. Everything worked properly prior to adding the new domain. I did add the new domain to the domain table, and have loaded the domain module with modparams as follows. modparam("domain", "db_url", "mysql://openser:mypass@myhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") Existing domain: sip1.mydomain.com New domain: sip1.mydomain.com user1: X-Lite (behind NAT) private-IP (192.168.x.x) Registers using domain sip1.mydomain.com user2: Linksys PAP2-NA public-IP (same subnet as Utsarcom) Registers using domain sip1.mydomain.com user3: Utstarcom iAN-02EX public-IP (63.225.x.x) Registers using domain sip2.mydomain.com Here is what happens now... - All users can register fine. - user1 & user2 can only receives calls from user3 (can't call others or pstn , 407 Proxy Auth Required) - user3 is the only one who can call others (calls to user1,user2, & pstn are fine) It's probably just something simple I missed, but I can't seem to figure it out. As always, thanks for your assistance. - Daryl

4 3

[Serusers] DNS problem
by Grabias Dariusz 15 Nov '05

15 Nov '05

Hello , I would like to use domain name of SER instead of IP adress during indetification I set dns=yes,rev_dns=yes and alias="sip.siptest" in ser.cfg but something strange has happened. Sip Client nr1(it's number 204) sends " INVITE sip:203@sip.siptest " to SER and SER sends "INVITE sip:203@192.168.1.204" to sip client nr2 (it's number 203). It looks very suspicious because SER should send " INVITE sip:204@sip.siptest " to client nr 2. It results in trouble when you try to phone from one client to another but reject the call on one client. One is disconnected and second one waits for a call . Could you advice something ??? Pozdrawiam / Best Regards Dariusz Grabias SIEMENS Sp. z o.o. Com SDC S&E C 5 ul. Strzegomska 52A, 53-311 Wroclaw, Poland tel +48 71 79 92 695 mobile: +48 xxx xxx xxx mailto:dariusz.grabias@siemens.com

2 1

[Serusers] Multiplexing RTP streams
by Noel Sharpe 15 Nov '05

15 Nov '05

Hi All I was wondering if any thought (effort) has been given to trying to combine multiple rtp streams that are going to the same destination to save on IP overhead from the many small packets. I guess this would be a use case for RTPProxy (or mediaproxy) or it could be implemented as an application infront of rtpproxy. I know a number of vendors of hardware gateways (quintum/mediatrix) do this, and report massive savings (50% if memory serves) of bandwidth. I doubt very much that they use an open standard, which means that integrating with these devices is impractical. To me it seems that this would be a very useful ability, especially when you are sending all your calls to a single remote gateway. The remote gatway would have to have similar software to de-multiplex the traffic on the other side, but if you had control of both ends of an expensive link, this would be very useful. Noel

1 0

[Serusers] Unable to stop SER: Need help
by Dhiman D Chowdhury 15 Nov '05

15 Nov '05

Hi all, I will appreciate your kind help. I am unable to stop ser and tried the followings: 1. service ser stop. [It failed but ser restart wroks] 2. killall ser. [ser: no process killed] Can anybody please help? Thanks Dhiman

4 3

[Serusers] how to make use of the function, is_from_local, in ser?
by zhangshuai 15 Nov '05

15 Nov '05

Dear all, We want to use multiple mediaproxy by using dispatcher. We set SIP_DOMIAN = mydomain.com, set up two user: A(a)mydomain.com & B(a)mydomain.com. They both register successfully. When user A & B dials for each other, the functions of is_from_local and is_destination_local both return False, so the dispatcher only return "warning: no proxy available for call: mydomain.com -> *.*.*.*" In mysql, the domain table is empty, too. How come? Thanks in advance. My ser config is as follow: debug=9 fork=no log_stderror=yes check_via=no dns=no rev_dns=no listen=218.97.252.38 listen=127.0.0.1 alias=sensates.com alias=218.97.252.38 port=5060 children=4 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/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" modparam("usrloc", "db_mode", 1) modparam("registrar", "use_domain", 1) modparam("auth_db", "calculate_ha1", 2) modparam("auth_db|domain|uri_db|usrloc","db_url","mysql://ser:heslo@localhost/ser") modparam("domain", "db_mode", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("mediaproxy", "natping_interval", 5) modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-asymmetric-clients") ............................. Best regards,

1 0

[Users] use of nathelper!
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all I want to use nathelper for nat transversal. But when i make a call, i have this error: 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: force_rtp_proxy2: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies 0(11117) ERROR: unforce_rtp_proxy: no available proxies What's matter? does i install a rtp proxy in a host? best regards! Serge

4 3

[Users] nat transversal
by messa＠innsof.com 15 Nov '05

15 Nov '05

Hi all Thanks for your help. I make a test in local (one xlite client calls another one, both in the same LAN than the openser) and everything goes well. But when i install openser in a machine with a public IP to make call between 2 clients in 2 differents LANs, i can't make a call. I don't know why? it is a client or an openser configuration problem? Can you help me? best regards!

2 4

[Users] modules_api
by Dariusz Dwornikowski 14 Nov '05

14 Nov '05

where can i find modules api for openser? -- Dariusz 'tdi' Dwornikowski mailto : tdi(a)vercom.pl gg: 2266034 JID : tdi(a)gentoo.pl

3 2

[Serusers] SEMS:announcement plug in
by rajesh.rajguru＠wipro.com 14 Nov '05

14 Nov '05

Hi, In our ser.cfg following is mentioned for announcement ---- if(!t_write_req("/tmp/am_fifo","announcement")) { t_reply("500","error contacting sems"); ----- we are using fifo for communicating between ser and sems. ser and sems are running on the same server (Fedora) When we start sems, the default announcement file mentioned in sems.conf (default_announce=default_en.wav) is not played. Also there is no error in /var/log/messages wrt announcement plug in What could be the problem ? How does sems decides which sound recorder to play ? The ll /tmp/am_fifo shows following prw-r--r-- 1 root root 0 Nov 14 09:48 /tmp/am_fifo Regards, Rajesh Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

2 1

[Serusers] prefix and the numeration plan
by Ldd 14 Nov '05

14 Nov '05

Hi, My carrier sent the information below, as installs? following prefix 45264 # and the numeration plan below: 45264# CC + AC + NNNNNNNN I try : if (uri=~"^sip:00[0-9].*@.*") { if (is_user_in("From", "ld")) { sl_send_reply("403", "Payment required"); break; }; setflag(1); prefix("45264#") rewritehostport("ip-server:5060"); if (!t_relay()) { sl_reply_error(); }; break; }; Thanks

1 0

[Users] substr in uri
by Iqbal 14 Nov '05

14 Nov '05

Hi I am doing the following and for some reason it doesn't seem to work (please ignore syntax) if (uri=0[1-9] { substr/^0/00/} and then if {uri=00) Now should the second if statement now match, or not, it my config it doesn't seem to. iqbal

2 4

[Users] Geting status and reason in failure_route
by Taras Bendik 14 Nov '05

14 Nov '05

Hi ----------------------------- failure_route[1] { if (t_check_status("408|500|503")) { t_reply("503", "Sorry"); } } ----------------------------- Is it posible to add loging in failure route, so User get 503, but I get acctual info on what happen, because $rs $rr work only in onreply_route, but I need them or any equivalent in failure_route? So I can log exactly what happen. Regards, Taras

2 2

[Users] $ruri
by Taras Bendik 14 Nov '05

14 Nov '05

Hi Im using avp_write("sip:123456@1.2.3.4:4321", "i:100"); avp_pushto("$ruri","i:100"); t_relay(); And this is sended to 1.2.3.4:4321 like : INVITE sip:123456@1.2.3.4:4321 SIP/2.0 And this works fine. However can I make it some how to generate message like this : INVITE sip:123456 SIP/2.0 ??? Or in this case I need to use : avp_write("sip:123456", "i:100"); avp_pushto("$ruri","i:100"); t_relay_to_udp("1.2.3.4","4321"); Regards, Taras

2 1

[Users] LCR documentation, examples, pointers?
by rushowr 14 Nov '05

14 Nov '05

Hello all, Hate to be a bother to the list, but the documentation for the LCR module (and googling afterwards) left me slightly puzzled. . . The way I read it, it seems like EVERY user on the system would have to have an entry in the gw table, in order to match the FROM URI and get a gateway prioritized.... Anyone have some light to shed on LCR? Maybe a solid doc somewhere? Other than that, please keep up the excellent work! O_SER is definitely a wonderful piece of work! SKM

3 2

[Serusers] ACC early_media issue ...
by Mamen xu 14 Nov '05

14 Nov '05

Hello, all: I am using SER interworking with Cisco26 as PSTN gateway. So far, everything goes OK but the Radius billing. There is only STOP Radius request with no START Radius request at first. I think it's due to 183 with SDP issue. But it seems early_media in ACC module didn't give it a remedy. BTW, START/STOP runs OK in intra-domain call..which including 180 without SDP in signaling loop. My ser.cfg is as below: modparam("acc", "log_level",1) modparam("acc","radius_flag",1) modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc",report_ack",0) modparam("acc",early_media",1) # failed_transaction,report_cancels,radius_missed_flag...[uncommented] Who can rescue me? Thanks.

1 0

[Serusers] Audio player for SEMS
by hemal.sheth＠wipro.com 14 Nov '05

14 Nov '05

We are trying to use SEMS along with SER in our lab setup We have configured a dedicated SEMS for one IP phone to start with And enabled the announcement plug-in But the announcement message is not played Where does SEMS take the default player from? Thanks Hemal Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin(a)wipro.com immediately and destroy all copies of this message and any attachments.

1 0

[Users] Re: nat transversal with mediaproxy
by Daryl Sanders 13 Nov '05

13 Nov '05

On 11/13/05, serge messa <sergepmessa(a)yahoo.fr> wrote: > Hi > > Thanks for your reply. must i make changes in the > mediaproxy.ini , the mediaproxy.cfg or openser.cfg > file? I had to configure mediaproxy.ini, but I am running mediaproxy on a few different, boxes. You might not if you are only running it local. Can anyone confirm this? > I want to know if my openser.cfg config file have any > problem? > > Have you use openser with the mediaproxy module? If > so, send me your openser.cfg file and tell me how you > make it work. I am currently testing the onsip.org ser.cfg, and it works fine with mediaproxy. If you have not done so you really should read the onsio.org getting started docs. The mediaproxy info was very helpful to me. > Ps: i have some problem with my email > messa(a)innsof.com. This is my alternate e-mail. > > best regards! > > Serge > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com >

1 0

[Serusers] Question Compiling acc.so
by Leo Papadopoulos 13 Nov '05

13 Nov '05

Dear serusers, I am running ser 0.9.3 I want to enable the accounting package to write to MySQL. I realize I need to recompile to do this. Can I just compile ONLY the acc and somehow install this acc module? If so, how? Is it just easier or better to recompile all of SER? ________________________________ Telecom CTO Leo Papadopoulos 10 Old Stone Court Ridgefield, CT 06877 Tel: 203-438-8117 Cell: 203-788-6364 E-mail: leo(a)telecomCTO.com Web site: www.telecomCTO.com

2 2

[Serusers] Re: NAT Problem and Stun Server
by Elangainesan Shanmugam 13 Nov '05

13 Nov '05

Dear serusers, I am running SER 9.4 . it works great. I am running SER outside on public address I am also having small problem behind NAT. To resolv this I am running Stun server - public address with Stun Client - private address ( behind NAT address ) The clients were not able to find its location. But If i do enter Public ip addresss for that system by finding via www.ip2location.com , Both incomming and out going works. The Problem is every time , i have to manually enter the ipaddress. Not sure, if I have to put both Stun server and Stun Client on the public network. And not sure, Stun Client process gives the ip address of the location Please help me Regards Nesan ----- Original Message ----- From: <serusers-request(a)lists.iptel.org> To: <serusers(a)lists.iptel.org> Sent: Sunday, November 13, 2005 5:00 AM Subject: Serusers Digest, Vol 31, Issue 15 > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. LCR question (Juha Heinanen) > 2. Question Compiling acc.so (Leo Papadopoulos) > 3. Re: NAT Traversal (Thomas Britis) > 4. Multi-domain setup problem (Daryl Sanders) > 5. Re: Question Compiling acc.so (sip) > 6. Re: mediaproxy for incoming (George Lambson) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 12 Nov 2005 15:48:01 +0200 > From: Juha Heinanen <jh(a)tutpro.com> > Subject: [Serusers] LCR question > To: Stefan Prelle <s.prelle(a)broadnet.de> > Cc: serusers(a)lists.iptel.org > Message-ID: <17269.62097.141780.968729(a)rautu.tutpro.com> > Content-Type: text/plain; charset=us-ascii > > Stefan Prelle writes: > > > I am having huge performance problems. The OpenSER drops to effectivly > > 1 CPS. My lcr table contains only 20 entries and I have just 2 gateways > > configured, so this shouldn't be a problem. > > currently load_gws() makes a complex mysql query, but only one mysql > query per second seems very slow to me. load_gws could be rewritten > totally in c, but so far nobody has had time to do so. > > -- juha > > > > ------------------------------ > > Message: 2 > Date: Sat, 12 Nov 2005 10:26:34 -0500 > From: "Leo Papadopoulos" <leo(a)telecomcto.com> > Subject: [Serusers] Question Compiling acc.so > To: <serusers(a)lists.iptel.org> > Message-ID: <006201c5e79d$77ea4250$0a01010a@leoathlon> > Content-Type: text/plain; charset="us-ascii" > > Dear serusers, > > I am running ser 0.9.3 > I want to enable the accounting package to write to MySQL. I realize I need > to recompile to do this. > > Can I just compile ONLY the acc and somehow install this acc module? If so, > how? > Is it just easier or better to recompile all of SER? > > ________________________________ > Telecom CTO > Leo Papadopoulos > 10 Old Stone Court > Ridgefield, CT 06877 > Tel: 203-438-8117 > Cell: 203-788-6364 > E-mail: leo(a)telecomCTO.com > Web site: www.telecomCTO.com > > > > > ------------------------------ > > Message: 3 > Date: Sat, 12 Nov 2005 09:36:54 -0200 > From: Thomas Britis <thomas(a)tcnet.com.br> > Subject: Re: [Serusers] NAT Traversal > To: "Greger V. Teigre" <greger(a)teigre.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <4375D3D6.1070805(a)tcnet.com.br> > Content-Type: text/plain; charset=us-ascii; format=flowed > > I tried. Everything goes fine until I change the default configuration > from onsip.org. All the configuration files are basical and needs a lot > of changes to make me happy, heh. > > If anyone has an idea, it would be great. > > Thank you. > > Greger V. Teigre wrote: > > Seems like you could use the Getting Started document from ONsip.org ;-) > > g-) > > ----- Original Message ----- From: "Thomas Britis" <thomas(a)tcnet.com.br> > > To: <serusers(a)lists.iptel.org> > > Sent: Friday, November 11, 2005 5:58 PM > > Subject: [Serusers] NAT Traversal > > > > > >> Hi, > >> > >> I'm working on a popular cenario (I think) of SER's implementation but > >> I'm having some difficult on making clients behind NAT to work well. > >> I tried to use rtpproxy and mediaproxy (not together, of course) but > >> without success. > >> Here is my scenario: SER with public IP address with rtpproxy and/or > >> mediaproxy up and running. Clients are behind NAT with NO firewall > >> blocking any ports. Or even one client behind NAT and other with valid > >> IP (by the way, all clients that are not behind nat works fine). The one > >> behind nat do not receive any voice but can send it. > >> > >> I'm attaching my ser.cfg (both, the rtpproxy and mediaproxy one) and > >> appreciate any help. > >> > >> Thank's in advance. > >> -- > >> Thomas Storino Britis > >> TCNet Informatica e Telecomunicacoes LTDA > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> # > >> # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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) > >> > >> # TCNet Configuration > >> # hostname matching an alias will satisfy the condition uri==myself. > >> listen="200.167.20.34" > >> alias="e-voip.com.br" > >> alias="200.167.20.34" > >> uid="ser" > >> gid="ser" > >> > >> 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:pass@localhost/ser" > >> > >> # Modules > >> loadmodule "/tcnet/ser/lib/ser/modules/domain.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/sl.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/tm.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/rr.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/mediaproxy.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/textops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/acc.so" > >> > >> # Modules parameter > >> # auth_db > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:pass@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> > >> # mediaproxy > >> modparam("mediaproxy", "mediaproxy_socket", > >> "/var/run/proxydispatcher.sock") > >> modparam("mediaproxy", "natping_interval", 30) > >> modparam("mediaproxy", "sip_asymmetrics", > >> "/tcnet/mediaproxy/sip-asymmetrics-clients") > >> modparam("mediaproxy", "rtp_asymmetrics", > >> "/tcnet/mediaproxy/rtp-asymmetrics-clients") > >> > >> # usrloc > >> modparam("usrloc", "db_mode", 2) > >> > >> # rr > >> modparam("rr", "enable_full_lr", 1) > >> > >> # registrar > >> modparam("registrar", "nat_flag", 6) > >> > >> # acc > >> modparam("acc", "radius_config", > >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") > >> modparam("acc", "log_level", 1) > >> modparam("acc", "log_flag", 1) > >> modparam("acc", "radius_flag", 1) > >> modparam("acc", "radius_missed_flag", 2) > >> modparam("acc", "log_fmt", "miocfst") > >> modparam("acc", "failed_transactions", 1) > >> > >> # tm > >> modparam("tm", "fr_inv_timer", 27) > >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") > >> > >> # main routing logic > >> route { > >> > >> if (method == "BYE" || method == "CANCEL") { > >> end_media_session(); > >> setflag(1); # accounting > >> setflag(2); # missed call > >> }; > >> > >> # initial sanity checks -- messages with > >> # max_forwards==0, or excessively long requests > >> if (!mf_process_maxfwd_header("10")) { > >> if (method!="ACK") { > >> sl_send_reply("483", "Too Many Hops"); > >> break; > >> }; > >> }; > >> if (msg:len > max_len) { > >> if (method!="ACK") { > >> sl_send_reply("513", "Message too big"); > >> break; > >> }; > >> }; > >> > >> if (method!="REGISTER") { > >> record_route (); > >> }; > >> > >> if (loose_route()) { > >> if (method=="BYE" || method=="CANCEL") { > >> end_media_session(); > >> setflag(1); > >> setflag(2); > >> }; > >> if (method=="INVITE" || method=="ACK") { > >> use_media_proxy(); > >> }; > >> route(1); > >> break; > >> }; > >> > >> if (uri == myself) { > >> if (method == "REGISTER") { > >> if (search("^(Contact|m): > >> .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { > >> log("LOG: alert: someone trying to set aor==contact\n"); > >> sl_send_reply("476", "No Server Address in Contacts Allowed"); > >> break; > >> }; > >> > >> route(2); # WWW authentication > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> break; > >> }; > >> > >> if (method == "INVITE") { > >> route(3); # PROXY authentication > >> if (uri=~"^sip:0") { > >> route(4); # Asterisk > >> }; > >> > >> consume_credentials(); > >> > >> #lookup("location"); > >> if (!lookup("location")) { > >> if (!lookup("aliases")) { > >> if (uri=~"^sip:[3789][0-9]") { > >> prefix("035"); > >> route(4); # Asterisk > >> } else { > >> sl_send_reply ("404", "Not Found"); > >> break; > >> }; > >> }; > >> }; > >> setflag(1); > >> setflag(2); > >> route(1); > >> break; > >> }; > >> }; > >> route(1); > >> } > >> > >> # Fix NATED client on reply route. > >> onreply_route[1] { > >> if (client_nat_test("1") && status=~"(183)|(2[0-9][0-9])") { > >> fix_contact(); > >> }; > >> use_media_proxy (); > >> } > >> > >> # Normal relay > >> route[1] { > >> > >> if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { > >> sl_send_reply("479", "We don't forward to private IP addresses"); > >> break; > >> }; > >> > >> t_on_reply("1"); > >> t_on_failure("1"); > >> > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> # WWW authentication > >> route[2] { > >> if (client_nat_test("3")) { > >> setflag(6); > >> force_rport(); > >> fix_contact(); > >> }; > >> > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("", "subscriber")) { > >> www_challenge("", "0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply ("401", "Unauthorized"); > >> break; > >> }; > >> > >> } > >> > >> # PROXY authentication > >> route[3] { > >> if (!proxy_authorize("", "subscriber")) { > >> proxy_challenge("", "0"); > >> break; > >> }; > >> } > >> > >> # Asterisk - E-Voip > >> route[4] { > >> t_on_reply("1"); > >> rewritehost("200.167.20.26"); > >> avp_write("i:45", "inv_timeout"); > >> } > >> > >> failure_route[1] { > >> end_media_session(); > >> } > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> # > >> # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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) > >> > >> # TCNet Configuration > >> # hostname matching an alias will satisfy the condition uri==myself. > >> listen="200.167.20.34" > >> alias="e-voip.com.br" > >> alias="200.167.20.34" > >> uid="ser" > >> gid="ser" > >> > >> 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:pass@localhost/ser" > >> > >> # Modules > >> loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/sl.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/tm.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/rr.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/textops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" > >> loadmodule "/tcnet/ser/lib/ser/modules/acc.so" > >> > >> # Modules parameter > >> # auth_db > >> modparam("auth_db|uri_db|usrloc", "db_url", > >> "mysql://ser:pass@localhost/ser") > >> modparam("auth_db", "calculate_ha1", 1) > >> modparam("auth_db", "password_column", "password") > >> > >> # nathelper > >> modparam("nathelper", "natping_interval", 30) > >> modparam("nathelper", "ping_nated_only", 1) > >> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") > >> > >> # usrloc > >> modparam("usrloc", "db_mode", 2) > >> > >> # rr > >> modparam("rr", "enable_full_lr", 1) > >> > >> # registrar > >> modparam("registrar", "nat_flag", 6) > >> > >> # acc > >> modparam("acc", "radius_config", > >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") > >> modparam("acc", "log_level", 1) > >> modparam("acc", "log_flag", 1) > >> modparam("acc", "radius_flag", 1) > >> modparam("acc", "radius_missed_flag", 2) > >> modparam("acc", "log_fmt", "miocfst") > >> modparam("acc", "failed_transactions", 1) > >> > >> # tm > >> modparam("tm", "fr_inv_timer", 27) > >> modparam("tm", "fr_inv_timer_avp", "inv_timeout") > >> > >> # main routing logic > >> route { > >> > >> if (method == "BYE" || method == "CANCEL") { > >> unforce_rtp_proxy(); > >> setflag(1); # accounting > >> setflag(2); # missed call > >> }; > >> > >> # 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 (method!="REGISTER") { > >> record_route (); > >> }; > >> > >> if (loose_route()) { > >> if (method == "BYE") { > >> setflag(1); > >> setflag(2); > >> unforce_rtp_proxy(); > >> }; > >> route(1); > >> break; > >> }; > >> > >> if (uri == myself) { > >> if (method=="INVITE") { > >> fix_nated_sdp ("1"); > >> }; > >> > >> if (method == "REGISTER") { > >> if (search("^(Contact|m): > >> .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { > >> log("LOG: alert: someone trying to set aor==contact\n"); > >> sl_send_reply("476", "No Server Address in Contacts Allowed"); > >> break; > >> }; > >> > >> route(2); # WWW authentication > >> > >> if (!save("location")) { > >> sl_reply_error(); > >> }; > >> break; > >> }; > >> > >> if (method == "INVITE") { > >> route(3); # PROXY authentication > >> if (uri=~"^sip:0") { > >> route(4); # Asterisk > >> }; > >> > >> consume_credentials(); > >> > >> #lookup("location"); > >> if (!lookup("location")) { > >> if (!lookup("aliases")) { > >> if (uri=~"^sip:[3789][0-9]") { > >> prefix("035"); > >> route(4); # Asterisk > >> } else { > >> sl_send_reply ("404", "Not Found"); > >> break; > >> }; > >> }; > >> }; > >> setflag(1); > >> setflag(2); > >> route(1); > >> break; > >> }; > >> }; > >> route(1); > >> } > >> > >> # Fix NATED client on reply route. > >> onreply_route[1] { > >> if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { > >> if (!search("Content-Length:[ ]*0")) { > >> force_rtp_proxy(); > >> }; > >> }; > >> } > >> > >> # Normal relay > >> route[1] { > >> > >> if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { > >> sl_send_reply("479", "We don't forward to private IP addresses"); > >> break; > >> }; > >> > >> t_on_reply("1"); > >> > >> if (isflagset(6) || isflagset(5)) { > >> log (1, "oi"); > >> force_rtp_proxy (); > >> }; > >> > >> if (!t_relay()) { > >> sl_reply_error(); > >> }; > >> } > >> > >> # WWW authentication > >> route[2] { > >> if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { > >> setflag(6); > >> fix_nated_register(); > >> force_rport(); > >> }; > >> > >> sl_send_reply("100", "Trying"); > >> > >> if (!www_authorize("", "subscriber")) { > >> www_challenge("", "0"); > >> break; > >> }; > >> > >> if (!check_to()) { > >> sl_send_reply ("401", "Unauthorized"); > >> break; > >> }; > >> > >> if (isflagset(5)) { > >> setflag(6); > >> }; > >> } > >> > >> # PROXY authentication > >> route[3] { > >> if (nat_uac_test("19")) { > >> setflag(6); > >> }; > >> > >> if (!proxy_authorize("", "subscriber")) { > >> proxy_challenge("", "0"); > >> break; > >> }; > >> } > >> > >> # Asterisk - E-Voip > >> route[4] { > >> if (isflagset(6)) { > >> force_rport(); > >> fix_nated_contact(); > >> force_rtp_proxy(); > >> }; > >> > >> t_on_reply("1"); > >> rewritehost("200.167.20.26"); > >> avp_write("i:45", "inv_timeout"); > >> } > >> > > > > > > -------------------------------------------------------------------------- ------ > > > > > > > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >> > > > > > > > > -- > Thomas Storino Britis > TCNet Informatica e Telecomunicacoes LTDA > > > > ------------------------------ > > Message: 4 > Date: Sat, 12 Nov 2005 14:59:01 -0700 > From: Daryl Sanders <daryl.sanders(a)gmail.com> > Subject: [Serusers] Multi-domain setup problem > To: users(a)openser.org, SER Users <serusers(a)lists.iptel.org> > Message-ID: > <cffd7ecd0511121359l3d7a135ev84f81087534e3a84(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Everyone, > > I just added a second domain to my openser box and screwed things up somehow. > Maybe someone can help figure out what's going on. > I'm using the new onsip config (unmodified except IPs & replacing > breaks w/returns). > I'm using mediaproxy for NAT. Everything worked properly prior to > adding the new domain. > I did add the new domain to the domain table, and have loaded the > domain module with modparams as follows. > > modparam("domain", "db_url", "mysql://openser:mypass@myhost/openser") > modparam("domain", "db_mode", 1) > modparam("domain", "domain_table", "domain") > modparam("domain", "domain_col", "domain") > > Existing domain: sip1.mydomain.com > New domain: sip1.mydomain.com > > user1: X-Lite (behind NAT) > private-IP (192.168.x.x) > Registers using domain sip1.mydomain.com > > user2: Linksys PAP2-NA > public-IP (same subnet as Utsarcom) > Registers using domain sip1.mydomain.com > > user3: Utstarcom iAN-02EX > public-IP (63.225.x.x) > Registers using domain sip2.mydomain.com > > Here is what happens now... > > - All users can register fine. > - user1 & user2 can only receives calls from user3 (can't call others > or pstn , 407 Proxy Auth Required) > - user3 is the only one who can call others (calls to user1,user2, & > pstn are fine) > > It's probably just something simple I missed, but I can't seem to > figure it out. As always, thanks for your assistance. > > - Daryl > > > > ------------------------------ > > Message: 5 > Date: Sat, 12 Nov 2005 17:02:34 -0500 > From: "sip" <sip(a)arcdiv.com> > Subject: Re: [Serusers] Question Compiling acc.so > To: "Leo Papadopoulos" <leo(a)telecomcto.com>, <serusers(a)lists.iptel.org> > Message-ID: <20051112215625.M80825(a)infinideas.com> > Content-Type: text/plain; charset=iso-8859-1 > > It's actually pretty easy to just recompile/install a module at a time. The > recompilation part is in the docs (INSTALL file). > > Basically, from the root directory of your source code... > > make modules=modules/acc modules > > That would recompile the acc module. My recommendation, however, is to go > INTO the src/modules/acc directoy and do a make clean (it will whine about how > you should run this from the root directory, but ignore that -- that's just a > stock message and doesn't pertain to make cleans). THEN recompile the module > from the root of the src tree to make sure any stale object files are no > longer lying about mucking up your recompile. > > Once it's been compiled, stop ser (important step), and just copy the > resulting .so (which will be in the modules/acc directory) file into the > /usr/local/lib/ser/modules directory and you're good to go. > > Restart ser and watch for weird error messages in the log. :) > > STOPPING ser is very important if you're replacing one of its modules, or it > will panic, die, and leave a bunch of core files sitting in your / tree. > Always messy. > > N. > > > On Sat, 12 Nov 2005 10:26:34 -0500, Leo Papadopoulos wrote > > Dear serusers, > > > > I am running ser 0.9.3 > > I want to enable the accounting package to write to MySQL. I realize > > I need to recompile to do this. > > > > Can I just compile ONLY the acc and somehow install this acc module? > > If so, how? Is it just easier or better to recompile all of SER? > > > > ________________________________ > > Telecom CTO > > Leo Papadopoulos > > 10 Old Stone Court > > Ridgefield, CT 06877 > > Tel: 203-438-8117 > > Cell: 203-788-6364 > > E-mail: leo(a)telecomCTO.com > > Web site: www.telecomCTO.com > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > ------------------------------ > > Message: 6 > Date: Sat, 12 Nov 2005 17:50:37 -0700 > From: George Lambson <LambsonGE(a)mtc.byu.edu> > Subject: Re: [Serusers] mediaproxy for incoming > To: greger(a)teigre.com > Cc: serusers(a)lists.iptel.org > Message-ID: <s3762bd8.008(a)mtcemail.mtc.byu.edu> > Content-Type: text/plain; charset=US-ASCII > > Any idea what I might be doing wrong? > > I have had some issues with setting up the served domain. I use x-lite for my SIP UAs and have to put the host name of my SER server to get them to log in, even though I set the environment variable for the SIP domain of the server to my domain. Do you think that it might be related? > > Thanks, > George > > >>> "Greger V. Teigre" <greger(a)teigre.com> 11/11/2005 12:19:59 AM >>> > I believe the standard NAT traversal configs from onsip.org should handle > that. > g-) > ----- Original Message ----- > From: "George Lambson" <LambsonGE(a)mtc.byu.edu> > To: <serusers(a)lists.iptel.org> > Sent: Friday, November 11, 2005 12:51 AM > Subject: [Serusers] mediaproxy for incoming > > > > Is anyone using mediaproxy for INCOMING calls? > > > > What I mean is: is SER capable of accepting INVITE messages from an > > unregistered UA that is behind a NAT Firewall and connecting them with a > > registered UA? > > > > I would like to be able to receive incoming calls for my domain from > > anyone on the internet and connect them to the proper user. Please tell me > > anyone if you are successfully doing this. > > > > Thanks, > > George > > > > _______________________________________________ > > 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 > > > End of Serusers Digest, Vol 31, Issue 15 > **************************************** >

1 0

Re: [Serusers] mediaproxy for incoming
by George Lambson 13 Nov '05

13 Nov '05

Any idea what I might be doing wrong? I have had some issues with setting up the served domain. I use x-lite for my SIP UAs and have to put the host name of my SER server to get them to log in, even though I set the environment variable for the SIP domain of the server to my domain. Do you think that it might be related? Thanks, George >>> "Greger V. Teigre" <greger(a)teigre.com> 11/11/2005 12:19:59 AM >>> I believe the standard NAT traversal configs from onsip.org should handle that. g-) ----- Original Message ----- From: "George Lambson" <LambsonGE(a)mtc.byu.edu> To: <serusers(a)lists.iptel.org> Sent: Friday, November 11, 2005 12:51 AM Subject: [Serusers] mediaproxy for incoming > Is anyone using mediaproxy for INCOMING calls? > > What I mean is: is SER capable of accepting INVITE messages from an > unregistered UA that is behind a NAT Firewall and connecting them with a > registered UA? > > I would like to be able to receive incoming calls for my domain from > anyone on the internet and connect them to the proper user. Please tell me > anyone if you are successfully doing this. > > Thanks, > George > > _______________________________________________ > 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

[Users] openser config file to use mediaproxy
by messa＠innsof.com 12 Nov '05

12 Nov '05

Hi all I want to use madiaproxy module to do nat transversal. But it not work for me. I run the media proxy server (proxy server) in the same host than openser on a host with public IP and i set the openser configuration file to above: # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu 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 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/openser_fifo" #loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" #loadmodule "/usr/local/lib/openser/modules/auth.so" #loadmodule "/usr/local/lib/openser/modules/auth_radius.so" #loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" #loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" #loadmodule "/usr/local/lib/openser/modules/uri_db.so" #loadmodule "/usr/local/lib/openser/modules/uri_radius.so" #modparam("auth_db|usrloc|uri_db", "db_url","mysql://ser:heslo@localhost/ser") #modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 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) route { #---------------------------------------------------------------- # Sanity Check Section #----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; #----------------------------------------------------------------- # Record Route Section #----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); }; #----------------------------------------------------------------- # Loose Route Section #----------------------------------------------------------------- if (loose_route()) { if (has_totag() && method=="INVITE") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); exit; }; #----------------------------------------------------------------- # Call Type Processing Section #----------------------------------------------------------------- if (uri==myself) { if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); }; } route[1] { #----------------------------------------------------------------- # Default Message Handler #----------------------------------------------------------------- t_on_reply("1"); 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(); # fix_nated_contact(); force_rport(); }; sl_send_reply("100", "Trying"); # if (!www_authorize("","subscriber")) { # www_challenge("","0"); # exit; # }; # if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # exit; # }; # consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { #----------------------------------------------------------------- # INVITE Message Handler #----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; # } else if (!check_from()) { # if(!check_from()) { # sl_send_reply("403", "Use From=ID"); # exit; # }; # consume_credentials(); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { force_rtp_proxy(); }; } else if (nat_uac_test("1")) { fix_nated_contact(); }; } What's wrong? your comments are welcome. Thanks Serge MESSA

2 1

[Serusers] NAT Traversal
by Thomas Britis 12 Nov '05

12 Nov '05

Hi, I'm working on a popular cenario (I think) of SER's implementation but I'm having some difficult on making clients behind NAT to work well. I tried to use rtpproxy and mediaproxy (not together, of course) but without success. Here is my scenario: SER with public IP address with rtpproxy and/or mediaproxy up and running. Clients are behind NAT with NO firewall blocking any ports. Or even one client behind NAT and other with valid IP (by the way, all clients that are not behind nat works fine). The one behind nat do not receive any voice but can send it. I'm attaching my ser.cfg (both, the rtpproxy and mediaproxy one) and appreciate any help. Thank's in advance. -- Thomas Storino Britis TCNet Informatica e Telecomunicacoes LTDA # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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) # TCNet Configuration # hostname matching an alias will satisfy the condition uri==myself. listen="200.167.20.34" alias="e-voip.com.br" alias="200.167.20.34" uid="ser" gid="ser" 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:pass@localhost/ser" # Modules loadmodule "/tcnet/ser/lib/ser/modules/domain.so" loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" loadmodule "/tcnet/ser/lib/ser/modules/sl.so" loadmodule "/tcnet/ser/lib/ser/modules/tm.so" loadmodule "/tcnet/ser/lib/ser/modules/rr.so" loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" loadmodule "/tcnet/ser/lib/ser/modules/auth.so" loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" loadmodule "/tcnet/ser/lib/ser/modules/uri.so" loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" loadmodule "/tcnet/ser/lib/ser/modules/mediaproxy.so" loadmodule "/tcnet/ser/lib/ser/modules/textops.so" loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" loadmodule "/tcnet/ser/lib/ser/modules/acc.so" # Modules parameter # auth_db modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:pass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # mediaproxy modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "sip_asymmetrics", "/tcnet/mediaproxy/sip-asymmetrics-clients") modparam("mediaproxy", "rtp_asymmetrics", "/tcnet/mediaproxy/rtp-asymmetrics-clients") # usrloc modparam("usrloc", "db_mode", 2) # rr modparam("rr", "enable_full_lr", 1) # registrar modparam("registrar", "nat_flag", 6) # acc modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) # tm modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # main routing logic route { if (method == "BYE" || method == "CANCEL") { end_media_session(); setflag(1); # accounting setflag(2); # missed call }; # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if (method!="ACK") { sl_send_reply("483", "Too Many Hops"); break; }; }; if (msg:len > max_len) { if (method!="ACK") { sl_send_reply("513", "Message too big"); break; }; }; if (method!="REGISTER") { record_route (); }; if (loose_route()) { if (method=="BYE" || method=="CANCEL") { end_media_session(); setflag(1); setflag(2); }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; route(1); break; }; if (uri == myself) { if (method == "REGISTER") { if (search("^(Contact|m): .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { log("LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed"); break; }; route(2); # WWW authentication if (!save("location")) { sl_reply_error(); }; break; }; if (method == "INVITE") { route(3); # PROXY authentication if (uri=~"^sip:0") { route(4); # Asterisk }; consume_credentials(); #lookup("location"); if (!lookup("location")) { if (!lookup("aliases")) { if (uri=~"^sip:[3789][0-9]") { prefix("035"); route(4); # Asterisk } else { sl_send_reply ("404", "Not Found"); break; }; }; }; setflag(1); setflag(2); route(1); break; }; }; route(1); } # Fix NATED client on reply route. onreply_route[1] { if (client_nat_test("1") && status=~"(183)|(2[0-9][0-9])") { fix_contact(); }; use_media_proxy (); } # Normal relay route[1] { if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; } # WWW authentication route[2] { if (client_nat_test("3")) { setflag(6); force_rport(); fix_contact(); }; sl_send_reply("100", "Trying"); if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!check_to()) { sl_send_reply ("401", "Unauthorized"); break; }; } # PROXY authentication route[3] { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; } # Asterisk - E-Voip route[4] { t_on_reply("1"); rewritehost("200.167.20.26"); avp_write("i:45", "inv_timeout"); } failure_route[1] { end_media_session(); } # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 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) # TCNet Configuration # hostname matching an alias will satisfy the condition uri==myself. listen="200.167.20.34" alias="e-voip.com.br" alias="200.167.20.34" uid="ser" gid="ser" 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:pass@localhost/ser" # Modules loadmodule "/tcnet/ser/lib/ser/modules/mysql.so" loadmodule "/tcnet/ser/lib/ser/modules/sl.so" loadmodule "/tcnet/ser/lib/ser/modules/tm.so" loadmodule "/tcnet/ser/lib/ser/modules/rr.so" loadmodule "/tcnet/ser/lib/ser/modules/maxfwd.so" loadmodule "/tcnet/ser/lib/ser/modules/usrloc.so" loadmodule "/tcnet/ser/lib/ser/modules/registrar.so" loadmodule "/tcnet/ser/lib/ser/modules/auth.so" loadmodule "/tcnet/ser/lib/ser/modules/auth_db.so" loadmodule "/tcnet/ser/lib/ser/modules/uri.so" loadmodule "/tcnet/ser/lib/ser/modules/uri_db.so" loadmodule "/tcnet/ser/lib/ser/modules/nathelper.so" loadmodule "/tcnet/ser/lib/ser/modules/textops.so" loadmodule "/tcnet/ser/lib/ser/modules/avpops.so" loadmodule "/tcnet/ser/lib/ser/modules/acc.so" # Modules parameter # auth_db modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:pass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") # nathelper modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # usrloc modparam("usrloc", "db_mode", 2) # rr modparam("rr", "enable_full_lr", 1) # registrar modparam("registrar", "nat_flag", 6) # acc modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) # tm modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # main routing logic route { if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); setflag(1); # accounting setflag(2); # missed call }; # 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 (method!="REGISTER") { record_route (); }; if (loose_route()) { if (method == "BYE") { setflag(1); setflag(2); unforce_rtp_proxy(); }; route(1); break; }; if (uri == myself) { if (method=="INVITE") { fix_nated_sdp ("1"); }; if (method == "REGISTER") { if (search("^(Contact|m): .*(a)(200\.167\.20\.34|evoip03\.tcnet\.com\.br|sip\.e-voip\.com\.br)")) { log("LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed"); break; }; route(2); # WWW authentication if (!save("location")) { sl_reply_error(); }; break; }; if (method == "INVITE") { route(3); # PROXY authentication if (uri=~"^sip:0") { route(4); # Asterisk }; consume_credentials(); #lookup("location"); if (!lookup("location")) { if (!lookup("aliases")) { if (uri=~"^sip:[3789][0-9]") { prefix("035"); route(4); # Asterisk } else { sl_send_reply ("404", "Not Found"); break; }; }; }; setflag(1); setflag(2); route(1); break; }; }; route(1); } # Fix NATED client on reply route. onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; } # Normal relay route[1] { if (uri =~ "[@:](192\.168\.|10\.|172\.16)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; t_on_reply("1"); if (isflagset(6) || isflagset(5)) { log (1, "oi"); force_rtp_proxy (); }; if (!t_relay()) { sl_reply_error(); }; } # WWW authentication route[2] { if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("", "subscriber")) { www_challenge("", "0"); break; }; if (!check_to()) { sl_send_reply ("401", "Unauthorized"); break; }; if (isflagset(5)) { setflag(6); }; } # PROXY authentication route[3] { if (nat_uac_test("19")) { setflag(6); }; if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; }; } # Asterisk - E-Voip route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); rewritehost("200.167.20.26"); avp_write("i:45", "inv_timeout"); }

2 2

[Serusers] avpops question
by Andy Pyles 12 Nov '05

12 Nov '05

hi I have an avpops question I have an avp object of boolean type called forwardcalls. My question is how do I check if a specific user has this enabled or not? I have the following logic: if (avp_db_load("$ruri/username", "s:callfwd")) { xlog("L_ERR", "call forward [%Tf] method <%rm> r-uri <%ru>\n"); if ( avp_check("forwarded_calls", "eq/0/gi" )) { log(1, "forward calls is disabled!\n\n"); break; } else { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; } }; I think my avp_check is wrong. Any ideas?

1 1

[Serusers] Foward Call to iConnectHere
by b in 12 Nov '05

12 Nov '05

I want to modify my ser.cfg so that if a user dials *5(then their iConnectHere username, password and the pstn phone number they want to call), the sip server will forward to information to iConnectHere, where the call is sent to the pstn phone number. Ex: If a user's iConnectHere username is 12345678 and their password is 1122 and they want to call the pstn number 9-999-999-9999, they will dial *512345678112299999999999 to call the desired pstn telephone number. How can I modify my ser.cfg to include this? (If you can, please include the code I can insert into ser.cfg or a sample ser.cfg file). Thanks. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] SER as an ALG. (now w/version info)
by Kyle Bresin 11 Nov '05

11 Nov '05

One quick addition, I'm using the latest versions of SER and mediaproxy that I could find: ser 0.9.4 and mediaproxy 1.4.2. Original problem follows: I have been trying to configure SER w/MediaProxy module/server to be a drop in NAT solution in front of asterisk. With the twist that I want to have asterisk continue acting as a registrar rather than SER. Outbound calls from natted clients works perfectly currently. The problem I'm having is with SER passing the REGISTERS through to asterisk. SER/MP isn't rewriting the address in the packets to go back through SER, but instead sets it to be the NAT'd IP of the client. So, when asterisk receives a call, and tries to send it to the external nat'd IP of the client, there is no open session from asterisk, so the firewall denies the request. What I need to happen, is for SER/MP to rewrite the packet with it's own IP, so that asterisk sends it back to SER, and SER forwards it back down to the client, from a source IP which has an open firewall session. Any help would be greatly appreciated, hopefully it's just a matter of me having an extraneous function call somewhere, or being ignorant of the right function to use. Or, in ascii diagram form: NAT_IP client --- EXTERNAL_FW_IP ----- SER_IP ----- ASTERISK SER is giving asterisk the EXTERNAL_FW_IP for Registers, and I want it to give Asterisk the SER IP. Something that would make SER rewrite *all* register packets to it's own IP would even work, since clients that don't need NAT won't be sent to SER in the first place. My ser.cfg and mediaproxy.ini follow. Lemme know if you need any more info from me. And extra thanks to anyone who has taken the time reading this, I truly appreciate it. kyle. # ### ser.cfg ### # # ----------- 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=5070 #children=4 fifo="/tmp/ser_fifo" #fifo_db_url="mysql://ser:ser/ser" # ------------------ 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/dbtext.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" loadmodule "/usr/local/lib/ser/modules/domain.so" # !!! Media Proxy loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 20) # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 (client_nat_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER fix_contact(); # Rewrite contact with source IP of signalling force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed # use_media_proxy(); }; # 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(); 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; }; route(1); } route[1] { 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)) { use_media_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(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { use_media_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (client_nat_test("1")) { fix_contact(); use_media_proxy(); }; } ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 127.0.0.1 portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct

1 0

[Serusers] SER as an ALG.
by Kyle Bresin 11 Nov '05

11 Nov '05

I have been trying to configure SER w/MediaProxy module/server to be a drop in NAT solution in front of asterisk. With the twist that I want to have asterisk continue acting as a registrar rather than SER. Outbound calls from natted clients works perfectly currently. The problem I'm having is with SER passing the REGISTERS through to asterisk. SER/MP isn't rewriting the address in the packets to go back through SER, but instead sets it to be the NAT'd IP of the client. So, when asterisk receives a call, and tries to send it to the external nat'd IP of the client, there is no open session from asterisk, so the firewall denies the request. What I need to happen, is for SER/MP to rewrite the packet with it's own IP, so that asterisk sends it back to SER, and SER forwards it back down to the client, from a source IP which has an open firewall session. Any help would be greatly appreciated, hopefully it's just a matter of me having an extraneous function call somewhere, or being ignorant of the right function to use. Or, in ascii diagram form: NAT_IP client --- EXTERNAL_FW_IP ----- SER_IP ----- ASTERISK SER is giving asterisk the EXTERNAL_FW_IP for Registers, and I want it to give Asterisk the SER IP. Something that would make SER rewrite *all* register packets to it's own IP would even work, since clients that don't need NAT won't be sent to SER in the first place. My ser.cfg and mediaproxy.ini follow. Lemme know if you need any more info from me. And extra thanks to anyone who has taken the time reading this, I truly appreciate it. kyle. # ### ser.cfg ### # # ----------- 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=5070 #children=4 fifo="/tmp/ser_fifo" #fifo_db_url="mysql://ser:ser/ser" # ------------------ 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/dbtext.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" loadmodule "/usr/local/lib/ser/modules/domain.so" # !!! Media Proxy loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 20) # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 (client_nat_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER fix_contact(); # Rewrite contact with source IP of signalling force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed # use_media_proxy(); }; # 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(); 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; }; route(1); } route[1] { 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)) { use_media_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(); }; } onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { use_media_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (client_nat_test("1")) { fix_contact(); use_media_proxy(); }; } ; ; Configuration file for MediaProxy ; [Dispatcher] start = yes socket = /var/run/proxydispatcher.sock ;group = ser defaultProxy = /var/run/mediaproxy.sock [MediaProxy] start = yes socket = /var/run/mediaproxy.sock ;group = ser ;listen = None ;allow = None ;proxyIP = 127.0.0.1 portRange = 35000:65000 ;TOS = 0xb8 ;idleTimeout = 60 ;holdTimeout = 3600 ;forceClose = 0 accounting = off [Accounting] user = dbuser password = dbpass host = dbhost database = radius table = radacct

1 0

RE: [Serusers] Multiple backup routes
by Bjorn Asmul 11 Nov '05

11 Nov '05

The LCR module is your best bet. -- Bjorn -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marcello Lupo Sent: Friday, November 11, 2005 6:45 AM To: serusers(a)lists.iptel.org Subject: [Serusers] Multiple backup routes Hi, if in a system i need to have more that one PSTN gateway to try for a destination in sequential order (note that gateways are dinamyc and may change call by call) and the order is important because them can be ordered for costs of the call from the less cost to the higher. There is a way for example to append all of them as branch and try them in sequence?? Otherwise i have to process them in a separate route block and have a temporary table in a database where take next gateways to try for that destination and that specific call, removing every cycle the used gateway. Any ideas to accomplish it in one step and let SER manage it?? Thanks, Bye, Marcello _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

2 1

Re: [Serusers] DID exchange or buy
by Rehan Ahmed 11 Nov '05

11 Nov '05

Available on Special Order, but not for exchange at this time. Rehan On 11/11/05, George Lambson <LambsonGE(a)mtc.byu.edu> wrote: > > Do you offer toll free #'s in Australia? > > George Lambson > Missionary Training Center > Provo, Utah > > >>> Rehan Ahmed <rehan33(a)gmail.com> 11/10/2005 3:11 PM >>> > We would be interested in numbers from ANYWHERE in the world, :) > How many can you give? > Minimum i would like to exchange is 50 number at a time. > Rehan > > On 11/11/05, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > > > > Dear Rehan, > > > > Are you interested in Romanian numbers also? > > > > Rehan Ahmed wrote: > > > Dear Leon, > > > > > > We would like to exchange numbers from you, which numbers do you have > > > avaialble ? > > > > > > We can exchange numbers with you from the us, uk, and australia at > this > > > time. > > > > > > We have 21 other countries also available for sale on www.didx.org<http://www.didx.org> > <http://www.didx.org> > > > <http://www.didx.org> (wholesale) and virtualphoneline.com<http://virtualphoneline.com> > <http://virtualphoneline.com> > > > <http://virtualphoneline.com> ( retail web site) > > > > > > Please check it out. > > > > > > Rehan > > > > > > > > > On 11/11/05, *Leon Sun* <leon.sun(a)keywestcommunications.com > > > <mailto:leon.sun@keywestcommunications.com>> wrote: > > > > > > Hi, all > > > > > > > > > > > > I believe most of you are in Telecom. I like to see if any one can > > > provide DID service over the world. If so, please let me know, we > > > like to buy DIDs from different countries. At beginning, we can > > > start with one number as testing. Furthermore, we can sign contract > > > and make payment for more DIDs you provide. > > > > > > > > > > > > Regards > > > > > > > > > > > > Leon Sun > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org <mailto:serusers@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] Compiling ser and mysql module on RH ES em64t
by Thomas Nilsen 11 Nov '05

11 Nov '05

Hello everyone! Im changing platforms and I have problem compiling ser for my new arch. I now have em64t servers with rh es. Actually the problem is compiling ser with mysql module. - Mysql em64t rpm installed from mysql site. - ser 0.9.4 sourcetree. When i compile everything works ok. But making mysql module fails. It complains about some relocation problem between 32 and 64 bit libs. I cant understand this since mysql is em64 version and so is ser. Off course bin version of ser does not work too. Here mysql.o complains about not finding mysqlclient.so, allthough it is there.. :-/ Anyone had luck with this? Probably some makefile tweaking to get it to use correct libs or something since this sweet rh has lib64 and some lib stuff cinfusing both me and ld :-) Would appriciate help! Cheers.. Thomas Nilsen

1 0

[Users] OpenSer as a UA?
by Chris Hastie 11 Nov '05

11 Nov '05

Hello I'm struggling to understand how all the bits of a VOIP system fit together and whether OpenSer can act as a UA and authenticate to a remote proxy. What I want to achieve is this. Incoming SIP calls are first routed to a SIP phone on the network. If this is unavailable, the call is then redirected to another proxy of a voip service provider with whom I have an accout, whence it should be routed to my mobile phone. I guess I can rewrite the call URI to mymobilenumber(a)proxy.voipprovider.tld, but I will need to authenticate to the proxy to authorise the call and have it charged to me. Can I do this with SER, or do I need something else (Asterix?) for this sort of function? -- Chris Hastie

2 1

[Users] which clients can i use for openser?
by messa＠innsof.com 11 Nov '05

11 Nov '05

Hi all I want to know which client i can use with openser and how i can configure it. Help me please! Regards! Serge MESSA OVONO

5 4

[Users] client configuration
by messa＠innsof.com 11 Nov '05

11 Nov '05

Hi Bogdan Thanks for your reply. I know Xlite! But tell me how i can add users in openser and how i must configure my xlite client. I want to know if in the outbound proxy, i must put the ip adress of the external part of the media proxy ( proxy server in the openser documentation). best regards Serge

2 2

[Serusers] Multiple backup routes
by Marcello Lupo 11 Nov '05

11 Nov '05

Hi, if in a system i need to have more that one PSTN gateway to try for a destination in sequential order (note that gateways are dinamyc and may change call by call) and the order is important because them can be ordered for costs of the call from the less cost to the higher. There is a way for example to append all of them as branch and try them in sequence?? Otherwise i have to process them in a separate route block and have a temporary table in a database where take next gateways to try for that destination and that specific call, removing every cycle the used gateway. Any ideas to accomplish it in one step and let SER manage it?? Thanks, Bye, Marcello

1 0

Re: [Users] How to add Route headers?
by Bogdan-Andrei Iancu 11 Nov '05

11 Nov '05

right...you need the new header on the top... I'm afraid there is no way to control the insertion point from the script (default is at the end)....... maybe having an extra parameter on the function to say top or bottom?.....not sure.... regards, bogdan Gustavo García Bernardo wrote: >Hi Bogdan, > >I had seen that function, but i think append_hf appends headers at the >bottom of the message, and I need to put those Route headers above any other >existing Route header. > >Any other solution? > >Thanks once more. > >regards >G. > >-----Mensaje original----- >De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Enviado el: jueves, 10 de noviembre de 2005 13:00 >Para: Gustavo García Bernardo >CC: users(a)openser.org >Asunto: Re: [Users] How to add Route headers? > >Hi Gustavo, > >if you have fixed content, it's even simpler. Use the textops module to >add the headers. See the append_hf() function: >http://www.openser.org/docs/modules/1.1.x/textops.html#AEN189 > >regards, >bogdan > >Gustavo García Bernardo wrote: > > > >>Hi Bogdan, thx for your response >> >>I need to put two fixed route headers, like these: >> >>Route: user(a)as.domain.com >>Route: user(a)cscf.domain.com >> >>Can I do that with AVPops module? I think is not suitable for this task. >> >>BR, >> >>G. >> >>-----Mensaje original----- >>De: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Enviado el: jueves, 10 de noviembre de 2005 0:28 >>Para: Gustavo García Bernardo >>CC: users(a)openser.org >>Asunto: Re: [Users] How to add Route headers? >> >>Hi Gustavo, >> >>you can generate the body of the Route header(s) into AVP(s) be loading >>them from database of by building them via avpops functions (like >>avp_printf()). >>then just set the AVP(s) into Route header(s) via the avp_pushto() >> >> >function: > > >>http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_pushto >> >>mostly depends of the content you want to push into the headers. >> >>regards, >>bogdan >> >>Gustavo García Bernardo wrote: >> >> >> >> >> >>>Hi, >>> >>>Is posible to add more route headers to a request before relaying it? >>> >>>I’m trying to implement the IMS mechanism before relaying a request to >>>an application server, which consists in adding two route headers. >>> >>>Thx in advance. >>> >>>G. >>> >>>------------------------------------------------------------------------ >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >> >> >> >> > > > >

5 7

RE: [Serusers] Test tool to generate 30 simultaneous SIP calls
by Rodríguez García, Marcos 11 Nov '05

11 Nov '05

Does this tool generate also RTP traffic in thoese simultaneous calls? Thanks, Marcos -----Mensaje original----- De: Atle Samuelsen [mailto:clona@cyberhouse.no] Enviado el: viernes, 11 de noviembre de 2005 11:11 Para: Rodríguez García, Marcos CC: serusers(a)lists.iptel.org Asunto: Re: [Serusers] Test tool to generate 30 simultaneous SIP calls signalling, you could use sipp.. /A * Rodr?guez Garc?a, Marcos <mrogarcia(a)indra.es> [051111 11:03]: > Do you know any test tool that I could use to generate 30 simultaneous SIP calls from one PC? > > > > Thanks in advance, > > Marcos Rodr?guez > > ------------------------------------------------------------------------------------------------------------------- > Este correo electr?nico y, en su caso, cualquier fichero anexo al mismo, contiene informaci?n de car?cter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgaci?n, copia o distribuci?n a terceros sin la previa autorizaci?n escrita de Indra. En el caso de haber recibido este correo electr?nico por error, se ruega notificar inmediatamente esta circunstancia mediante reenv?o a la direcci?n electr?nica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTR?NICO CONSIDERE SU APORTACI?N A LA CONSERVACI?N DEL MEDIO AMBIENTE POR LA REDUCCI?N DE CONSUMO DE PAPEL. > > The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL. > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers ------------------------------------------------------------------------------------------------------------------- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de Indra. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTRÓNICO CONSIDERE SU APORTACIÓN A LA CONSERVACIÓN DEL MEDIO AMBIENTE POR LA REDUCCIÓN DE CONSUMO DE PAPEL. The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL.

2 1

[Users] Registrar-initiated Route changes on Invite?
by Joachim Fabini 11 Nov '05

11 Nov '05

Hi, We'd like to force the following SIP INVITE routing scenario: UA1->P1->P2->R->P2->P1->UA2. Unfortunately the Route is not known apriori and must be built dynamically. P1 knows the Route from P1 to R, R knows the Route from R to P1/UA2. The INVITE sent by UA1 is loosely routed to R based on the Route (Route: <sip:P2;lr>,<sip:R;lr>) that is added by P1 to UA1's INVITE. This works. The message arrives correctly to the registrar, Route header contains Route: <sip:R;lr> as it should. What we'd like to do: 1) Lookup the contact (lookup("location");) 2) Rewrite the Route field to the value Route: <sip:P2;lr>,<sip:P1;lr> that is stored by R in an AVP. 3) Forward the Invite on this route to UA2 Can someone please give some hints on how this is typically done in OpenSER? Although we rewrite the Invite's Route header field (new value stored in an AVP) at the Registrar, loose_route() does not work. We tried several alternatives including re-writing the Route header, pushing hard-coded strings into the Route, etc. but none of these solutions does what we want. What does work is if R self-forwards the Invite and only then adds a new Route header. Apparently loose_route() in R acts on the route header of the incoming INVITE - if present - and ignores route header changes done within the script. Thanks in advance --Joachim

5 8

[Users] xlog and request mothod in responses
by Klaus Darilion 11 Nov '05

11 Nov '05

Hi! I use xlog to log certain responses. I want to log the request method of a response. Using $rm in the reply prints <null>. Using $cs prints only the CSeq number, but no the method. Is there a way to achieve this? regards klaus

4 4

[Users] Help with NATHelper module and fix_nated_contact() function
by Anthony Chapellier 11 Nov '05

11 Nov '05

Hi, NatHelper documentation says fix_nated_contact is rewriting contact header with request source address:port but it seems the function doesn't work properly with my SER config... Indeed, when you analyze a SIP Request Packet you have the following : Message Header : Contact : <sip:352621@192.170.10.32:5060> #192.170.10.32:5060 is the private IP/Port and if this function was working properly, I understand you should get the following : Message Header : Contact: <sip:352621@80.114.25.8:2014> #80.114.25.8:2014 is the public IP/Port But the SIP packets that are getting out from SER Proxy to SIP Registrar Server have not their contact header field modified with public IP/Port... So, is the function not working ? or something else ? Thanks,

3 2

[Users] How to change contact
by Taras Bendik 11 Nov '05

11 Nov '05

Hi I need something similar so fix_nated_contact but with parametars so I can set up my own values for host:port. Can I do this in OpenSer? Can avp_pushto do this? Regards, Taras

2 1

[Serusers] scope & database operations for avp_write
by Ramin Nikaeen 11 Nov '05

11 Nov '05

Valued Colleagues, Can anyone tell me what is the scope of avp_write?! Is it global / per-transaction / per-message?! I would like to have something similar to it with global scope so that I can set the string "callfwd" once by executing: avp_write("sip:211@10.1.10.22:5060", "s:callfwd"); and then keep using it for each SIP message processing using: avp_pushto("$ruri", "s:callfwd"); Also, do the avp_write and avp_pushto involve Database access operations?! i.e. Does avp_write store the string in database and avp_pushto looks it up form the database or is the information stored in memory?! I want to avoid database access to ensure improved performance. Thanks ramin

2 1

[Serusers] jabber and pa modules
by Matteo Piazza 11 Nov '05

11 Nov '05

Hi, I have this problem with the jabber and PA mudules, I use Jive messenger as Jabber server on the same machine Someone can me help? ========================= log debug ========================= 0(15774) parse_headers: flags=-1 0(15774) check_via_address(192.168.9.181, 192.168.9.181, 0) 0(15774) REGISTER received -> reply okay 0(15774) XJAB: Going OFFline in Jabber network!!! 0(15774) XJAB: go offline in Jabber network 0(15774) parse_headers: flags=12 0(15774) DEBUG: add_param: tag=67b87d 0(15774) end of header reached, state=29 0(15774) XJAB:xj_wlist_check: entry does not exist for <sip:308@192.168.9.97> 0(15774) XJAB:xjab_manage_sipmsg: no open Jabber session for <sip:308@192.168.9.97>! 0(15774) parse_headers: flags=-1 =========================== ser.cfg ========================== modparam("jabber","db_url","mysql://root:****@localhost/sip_jab") modparam("jabber","jaddress","localhost") 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","192.168.9.97") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; # 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") { if (t_newtran()) { handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 2

[Serusers] prob jabber xml version
by Matteo Piazza 11 Nov '05

11 Nov '05

I try to use jabber module, but the jabber server send me this string: <?xml version='1.0' encoding='UTF-8'?><stream:stream xmlns:stream="http://etherx.jabber.org/streams" xmlns="jabber:client" from="panoramix.create-net.it" id="2abf6ae" xml:lang="en"> The jabber module want a string as the follow: <?xml version='1.0'?>.......... Any idea? -- ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

2 1

[Serusers] Accounting Problem
by Rommel Catabian 11 Nov '05

11 Nov '05

Im very new to SER. My colleague who just left the company already set up the system. We can now call sip-to-sip and sip-to-pstn. However, our problem is that we dont have accounting record for the pstn calls. For sip-to-sip, accounting is OK. >From the radius log of the pstn calls, I can see only STOP request but no START request. My question: What triggers the SER to send START accounting request to the RADIUS? All traffic to pstn goes to a gateway operated by a third party service provider. We are using freeradius. Thanks, Rommel __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

2 1

Re: [Users] enable NAT OPTION ping
by Bogdan-Andrei Iancu 11 Nov '05

11 Nov '05

Hi Raymond, to you experience the same behaviour if you disable SIP ping (by not setting flag 5) and falling back to UDP ping? regards, bogdan Raymond Chen wrote: >Hi Bogdan, > >No, registration happens every minute. > >Thanks > >Ray > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: 2005年11月10日 23:01 >To: Raymond Chen >Cc: users(a)openser.org >Subject: Re: [Users] enable NAT OPTION ping > >Hi Raymond, > >maybe the registration expires - check if when the PING stops, the >registered contact is still valid (not expired). > >regards, >bogdan > >Raymond Chen wrote: > > > >>Dear all, >> >>I want to force openser to ping all registered users with OPTIONS >>message, my configuration as follow, flag in location table is 512 for >>sip_nat_flag, and 513 for NAT. I can see the OPTIONS message send to >>the users and got replied for a few minutes, after that, it stop >>sending. Is there something I’ve done wrong? >> >>Thanks >> >>Ray >> >>modparam("registrar", "nat_flag", 6) >> >>modparam("registrar", "sip_natping_flag", 5) >> >>modparam("nathelper", "ping_nated_only", 0) >> >>modparam("nathelper", "natping_interval", 30) >> >>modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") >> >>modparam("nathelper", "sipping_method", "OPTIONS") >> >>modparam("nathelper", "sipping_from", "sip:keepitopen@xxxxxx.net") >> >>if (nat_uac_test("31")) { >> >>if (method == "REGISTER" || !search("^Record-Route:")) { >> >>fix_nated_contact(); # Rewrite contact with source IP of signalling >> >>if (method=="REGISTER") { >> >>fix_nated_register(); >> >>} else if (method == "INVITE") { >> >>fix_nated_sdp("1"); # Add direction=active to SDP >> >>}; >> >>force_rport(); # Add rport parameter to topmost Via >> >>append_hf("P-hint: Proxied\r\n"); >> >>setflag(6); # Mark as NATed >> >>}; >> >>}; >> >>if (uri==myself) { >> >>if (method=="REGISTER") { >> >>if (!www_authorize("", "subscriber")) { >> >>www_challenge("", "1"); >> >>exit(); >> >>}; >> >>Setflag(5); >> >>save("location"); >> >>return; >> >>}; >> >>alias_db_lookup("aliases"); >> >>if (lookup("location")) { >> >>append_hf("P-hint: usrloc applied\r\n"); >> >>route(1); >> >>return; >> >>}; >> >>}; >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> > > > > > > >

1 0

Re: [Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

1 0

[Serusers] Test tool to generate 30 simultaneous SIP calls
by Rodríguez García, Marcos 11 Nov '05

11 Nov '05

Do you know any test tool that I could use to generate 30 simultaneous SIP calls from one PC? Thanks in advance, Marcos Rodríguez ------------------------------------------------------------------------------------------------------------------- Este correo electrónico y, en su caso, cualquier fichero anexo al mismo, contiene información de carácter confidencial exclusivamente dirigida a su destinatario o destinatarios. Queda prohibida su divulgación, copia o distribución a terceros sin la previa autorización escrita de Indra. En el caso de haber recibido este correo electrónico por error, se ruega notificar inmediatamente esta circunstancia mediante reenvío a la dirección electrónica del remitente. POR FAVOR, ANTES DE IMPRIMIR ESTE CORREO ELECTRÓNICO CONSIDERE SU APORTACIÓN A LA CONSERVACIÓN DEL MEDIO AMBIENTE POR LA REDUCCIÓN DE CONSUMO DE PAPEL. The information in this e-mail and in any attachments is confidential and solely for the attention and use of the named addressee(s). You are hereby notified that any dissemination, distribution or copy of this communication is prohibited without the prior written consent of Indra. If you have received this communication in error, please, notify the sender by reply e-mail. PLEASE CONSIDER YOUR ENVIRONMENTAL RESPONSIBILITY BEFORE PRINTING THIS E-MAIL.

2 1

[Serusers] Call forwarding problem
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

1 0

[Users] UAC problem. UAC don't send final ACK if UAS send OK
by jltorre＠yahoo.com 11 Nov '05

11 Nov '05

uac module configuration is by default without modparam, and the problem is when the callee respond the call, the caller hear ring sound and callee voice, both of then, and UAC ( caller) don't do final ACK if reveive OK from UAS, because OK trace from UAS may be not correct. We only use uac_replace_from("","usuario(a)domain.com") for INVITE,ACK,CANCEL or BYE method in route section.We have use openser release 1.0.0 Have you a solution or is it a bug ? Thank in advance Jose __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com

2 3

Re: [Serusers] Calls from UAs behind NAT getting a 503 error...
by Klaus Darilion 11 Nov '05

11 Nov '05

Kyriakos wrote: > So Klaus, basically what i need is an extra handler for ACK? > > route[6] { > > # ------------------------------------------------------------------------ > # ACK Handler > # ------------------------------------------------------------------------ > if (method=="ACK") { > t_relay(); > return; > }; That should work > Does this affect the call flow? I mean Ive read that responses to ACK > requests are not that important. per standard: there is NEVER a response to an ACK > Im a newbie in SER and sip so please bare with me. > :-) please always cc: to the list. regards klaus > By the way thanks for the reply. > > Kyriakos > > >

3 2

[Serusers] Expire for User agent question
by rpagquil＠philonline.com 11 Nov '05

11 Nov '05

2 1

[Serusers] SER and LDAP
by Felipe Louback 11 Nov '05

11 Nov '05

Is there any way to authenticate SER users in LDAP without the use of RADIUS? If there is, could anyone point out where I could find information about it? I've searched some documents but found nothing about authentication with LDAP without using RADIUS.... Regards, Felipe -- Master Student - Electrical Engineering Department Computer Engineering and Telecommunications Research Group Universidade Federal de Minas Gerais - Brazil "Humanly speaking it is impossible; but with God anything is possible!" Jesus Christ in Matthews 19:26

4 11

[Serusers] question about relay
by yserge 11 Nov '05

11 Nov '05

Hi, I am using SER as a an entry point for the requests from the network and every request from the network is being relayed to another ( clusterized ) SIP server ( not a SER ) with multiple IP addresses in DNS. The server is running on 4 machines, any number of which can be down at any moment of time. Suppose, the DNS name is "sip.myserver.com". The incoming requests from the network have SER as host:port of RURI.. I was trying to relay the request using t_relay_to_udp("sip.myserver.com", "5060"), but all requests would always go to the same IP. A question: Is there any way to make SER relay requests to those IPs in round-robin manner? Or can I force SER to make DNS requests before processing each request? Another problem related to this situation, if the SER relays all requests to the same IP, and the server represented by that IP goes down, I have 100% call failure. Is there any way to avoid that? thanks, Sergei

2 1

[Serusers] mediaproxy for incoming
by George Lambson 11 Nov '05

11 Nov '05

Is anyone using mediaproxy for INCOMING calls? What I mean is: is SER capable of accepting INVITE messages from an unregistered UA that is behind a NAT Firewall and connecting them with a registered UA? I would like to be able to receive incoming calls for my domain from anyone on the internet and connect them to the proper user. Please tell me anyone if you are successfully doing this. Thanks, George

2 1

[Serusers] can't connect to RTP proxy
by lo ™ 11 Nov '05

11 Nov '05

hi all, i'm quite new to sip and i'm trying to setup ser with nat support. i read about stun but to rewrite contact with received and rport tags i guess i need nathelper module that, in turn, needs rtpproxy, doesn't it? i got ser-0.9.0, i downloaded rtpproxy from cvs.ser.berlios.de<http://cvs.ser.berlios.de>and i use the same nathelper module configuation and nat routing you can find here: http://www.informatik.uni-bremen.de/~prelle/terena/cookbook/main/ch04s07.ht… unlucky ser.log tells me: ERROR: send_rtpp_command: can't connect to RTP proxy WARNING: rtpp_test: can't get version of the RTP proxy WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily i've been searching google and mail.iptel.org <http://lists.iptel.org> there is a very similar thread: http://lists.iptel.org/pipermail/serusers/2005-July/thread.html#22154 but i'm forced to ask my own question, hoping someone can help me. maybe i just need to compile rtpproxy with particular parameters... but i couldn't find enough documentation about it. thank you so much!!

4 3

RE: [Serusers] Accounting Problem
by Rommel Catabian 11 Nov '05

11 Nov '05

Thanks for the immediate reply. We have this on the file ser.cfg: modparam("acc", "radius_flag", 2) setflag(2); ---->>> could this be on the wrong location? To route traffic to PSTN, we have this on the ser config: ----- if (uri=~"^sip:90087(.+)@.*") { rewritehostport("XXX.XXX.XXX.XXX:5060"); forward(uri:host, uri:port); exec_msg('(/usr/local/etc/ser/logs.csh)'); break; ------ Here is our ser.cfg file: -------------------------- debug=3 fork=yes log_stderror=yes check_via=no dns=no rev_dns=no port=5060 children=4 fifo="/tmp/ser_fifo" fifo_db_url="mysql://XXXX:XXXX@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" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/exec.so" modparam("usrloc", "db_mode", 2) # 0 Disable Authentication ; 2 Enable Authentication modparam("usrloc", "db_url", "mysql://XXXX:XXXX@localhost/ser") modparam("auth_db", "db_url", "mysql://XXXX:XXXX@localhost/ser") modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("acc", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) 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; }; if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { fix_nated_contact(); if (method == "INVITE") { fix_nated_sdp("1"); }; force_rport(); setflag(6); }; }; lookup("aliases"); if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; setflag(2); setflag(5); if (method=="INVITE") { exec_msg('(/usr/local/etc/ser/record.pl)'); }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri=~"^sip:90087(.+)@.*") { rewritehostport("202.64.248.209:5060"); forward(uri:host, uri:port); exec_msg('(/usr/local/etc/ser/logs.csh)'); break; }; #--------- IVR Start ----------- if (uri=~"^sip:222@.*") { if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; if (method=="ACK"){ t_relay(); break; }; t_reply("100","Trying -- just wait a minute !"); if(!t_write_unix("/tmp/am_sock","ivr")) { t_reply("500","error contacting sems"); }; break; }; #--------- IVR End ----------- if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("sip.easy.net.ph","subscriber")) { www_challenge("sip.easy.net.ph", "0"); break; }; save("location"); break; }; if (method=="INVITE") { exec_msg('(/usr/local/etc/ser/record.pl)'); }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (method=="INVITE") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="CANCEL") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="BYE") { exec_msg('/usr/local/etc/ser/record.pl'); }; if (method=="ACK") { exec_msg('/usr/local/etc/ser/record.pl'); }; setflag(2); 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 (isflagset(6)) { force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); } else if (nat_uac_test("1")) { fix_nated_contact(); }; } --- Matt Simpson <msimpson(a)commpartners.us> wrote: > > >My question: > >What triggers the SER to send START accounting > request > >to the RADIUS? > > > >Thanks, > > > >Rommel > > Rommel, there should be a statement at the top like: > modparam("acc","radius_flag",1) > > The "1" is the flag to account for a call in radius > > Then you would use: > setflag(1); > > to account for the call. It is hard to tell you > where to put the > statement without seeing your conf but that should > help. > > __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

1 0

[Serusers] SER between softswitches
by Matt Simpson 11 Nov '05

11 Nov '05

I'm using SER to shuttle calls between softswitches. I have "softswitch A" sending calls to the SER proxy, which then does an enum lookup to make a routing decision and then passes the call on to the appropriate softswitch. I have been having some strange problems with the setup and some questions about my config. I sometimes get: Nov 10 21:14:01 cpser ser[22087]: 483 Too many Hops Nov 10 21:14:01 cpser ser[22087]: Warning: sl_send_reply: I won't send a reply for ACK!! This mostly happens after I do a serctl restart to reload my config. I assume that is because ACKs are coming in that the stateful transaction processor doesn't know where to forward them to since I restarted? There's no other way to reload a config other than a full restart, correct? Also I see a lot of "408 Request Timeout" from far end gateways. Is there something wrong with my config? Here is my config: modparam("acc","radius_flag",1) modparam("acc","failed_transactions",0) modparam("tm", "fr_inv_timer", 5 ) route{ if ( !mf_process_maxfwd_header("10") ) { xlog("L_ERR","483 Too many Hops"); sl_send_reply("483","Too Many Hops"); drop(); }; if (msg:len >= 2048 ) { xlog("L_ERR","513 Message too Big"); sl_send_reply("513", "Message too big"); break; }; setflag(1); if (loose_route()) { t_relay(); break; }; if (method =="BYE") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="CANCEL") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="ACK") { if ( !t_relay()) { sl_reply_error(); } break; }; if (method =="INVITE") { record_route(); if (uri=~"^sip:[2-9]+@*") { prefix("+"); if ( !enum_query("e164.internal.net")) { strip(1); prefix("1"); rewritehost("192.168.10.5"); xlog("L_ERR","NANPA number to %tu from %fu carrier %ru\n"); t_on_failure("1"); t_relay(); break; } xlog("L_ERR","NANPA-ENUM number to %tu from %fu carrier %ru\n"); t_on_failure("1"); t_relay(); }; # end if block method = invite } } failure_route[1] { if(t_check_status("501|502|503|504|505|513")) { setflag(1); revert_uri(); rewritehost("192.168.50.3"); append_branch(); xlog("L_ERR","FAIL-5XX number to %tu from %fu carrier %ru\n"); t_relay(); } }

1 0

[Serusers] ACK Message problem
by Eddie Chan 11 Nov '05

11 Nov '05

Hi, I was making test call from a UA1(10.0.2.12) via SER (192.168.168.87) to another UA2 (192.168.168.33) where UA1 is a program written by us and UA2 is a softphone. When I cancel the call, I am having problem sending an Ack to SER. In the trace attached, when UA1 send a Cancel, it replies with 200 canceling in response to this Cancel, and 487 in response to the original INVITE message. Then UA1 send back an ACK in response to 487 but SER keep resending 487 message. It seems like SER doesn't recognize the ACK message for some reasons. Could anyone tell me what is the problem with this ACK message please? Below are the only errors I can see in the /var/adm/message when I start up ser. There is no error message during the test call. Nov 11 09:07:12 TakeAway ./ser[5557]: [ID 212014 daemon.warning] Warning: uac does not spread across the whole hash table Nov 11 09:07:12 TakeAway ./ser[5557]: [ID 499528 daemon.notice] Maxfwd module- initializing Nov 11 09:07:12 TakeAway ./ser[5558]: [ID 827449 daemon.warning] WARNING: no fifo_db_url given - fifo DB commands disabled! OS version: Sparc Solaris 5.9 Ser version: ser 0.9.3 Thanks, Eddie

1 0

[Serusers] rewrite contact header of INVITE message
by George Lambson 10 Nov '05

10 Nov '05

Hi all, How do I have SER rewrite the contact header of an INVITE message? If I do that, will mediaproxy work for the session? Thanks in advance, George

1 0

[Serusers] ACK Message problem
by Eddie Chan 10 Nov '05

10 Nov '05

Hi, I was making test call from a UA1(10.0.2.12) via SER (192.168.168.87) to another UA2 (192.168.168.33) where UA1 is a program written by us and UA2 is a softphone. When I cancel the call, I am having problem sending an Ack to SER. In the trace attached, when UA1 send a Cancel, it replies with 200 canceling in response to this Cancel, and 487 in response to the original INVITE message. Then UA1 send back an ACK in response to 487 but SER keep resending 487 message. It seems like SER doesn't recognize the ACK message for some reasons. Could anyone tell me what is the problem with this ACK message please? Below are the only errors I can see in the /var/adm/message when I start up ser. There is no error message during the test call. Nov 11 09:07:12 TakeAway ./ser[5557]: [ID 212014 daemon.warning] Warning: uac does not spread across the whole hash table Nov 11 09:07:12 TakeAway ./ser[5557]: [ID 499528 daemon.notice] Maxfwd module- initializing Nov 11 09:07:12 TakeAway ./ser[5558]: [ID 827449 daemon.warning] WARNING: no fifo_db_url given - fifo DB commands disabled! OS version: Sparc Solaris 5.9 Ser version: ser 0.9.3 Thanks, Eddie

1 0

Re: [Serusers] DID exchange or buy
by Rehan Ahmed 10 Nov '05

10 Nov '05

We would be interested in numbers from ANYWHERE in the world, :) How many can you give? Minimum i would like to exchange is 50 number at a time. Rehan On 11/11/05, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > > Dear Rehan, > > Are you interested in Romanian numbers also? > > Rehan Ahmed wrote: > > Dear Leon, > > > > We would like to exchange numbers from you, which numbers do you have > > avaialble ? > > > > We can exchange numbers with you from the us, uk, and australia at this > > time. > > > > We have 21 other countries also available for sale on www.didx.org<http://www.didx.org> > > <http://www.didx.org> (wholesale) and virtualphoneline.com<http://virtualphoneline.com> > > <http://virtualphoneline.com> ( retail web site) > > > > Please check it out. > > > > Rehan > > > > > > On 11/11/05, *Leon Sun* <leon.sun(a)keywestcommunications.com > > <mailto:leon.sun@keywestcommunications.com>> wrote: > > > > Hi, all > > > > > > > > I believe most of you are in Telecom. I like to see if any one can > > provide DID service over the world. If so, please let me know, we > > like to buy DIDs from different countries. At beginning, we can > > start with one number as testing. Furthermore, we can sign contract > > and make payment for more DIDs you provide. > > > > > > > > Regards > > > > > > > > Leon Sun > > > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org <mailto:serusers@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] Mediaproxy module and SIP over TCP
by HERNAN HEVIA 10 Nov '05

10 Nov '05

Hello, I've found the following issue: When my gateway (it has a public IP address) sends an Invite to SER over a TCP connection, the function client_nat_test(2) always returns true, even though the VIA HF ahs the gateway public IP address. Any idea of why it is doing so? Does it have to do with port comparisson? I tried to fix it using sip_asymmetrics file, but it didn't work. Any help on this file syntax? Thank you very much in advance Hernán

1 0

[Users] Thanks for the help
by Krishna Chava 10 Nov '05

10 Nov '05

Hi everyone, Thanks for your comments.. i had a working openser now thanks and am having registrations working.. Will post my problems if i had any while going through... thanks krishna

1 0

[Serusers] SER + SipXchange
by Kathleen McCallum 10 Nov '05

10 Nov '05

Hello everyone, I must use SER as a proxy server to send every request to SipXchange (except the MESSAGE request that will be handled by SER) Can I get some help on this?, Is there any example I could use as a reference?. The user must be registered in SipXchange (in SER it would be optional) If the request is "MESSAGE" then SER will handle the request. This is what I have done if (!method=="MESSAGE") { xlog("L_INFO","Foward process\n"); forward(SipXchangeServer,SipXchangeServerPort); break; } But I think something is missing, Kat.

1 0

[Serusers] replace function
by Xavier TRENTIN 10 Nov '05

10 Nov '05

Hi, I use ser 0.10.99+cvs20050428. When I use replace("foo","bar") in ser.cfg, is it supposed to replace all occurences of "foo" in the headers/body of the SIP message? It seems that it only replace the first occurence. Moreover, when I write : replace("foo","bar"); replace("foo","bar"); in my ser.cfg, it replace "foo" with "barbar". B.R. Xavier.

1 0

[Users] roadmap to v1.1.0
by Daniel-Constantin Mierla 10 Nov '05

10 Nov '05

Hi everybody, we are trying to collect the items to be included in the roadmap to the next release. I have written a draft with the ideas discussed on the mailing list, the items which were not accomplished in v1.0.0 and a few more. There is no weight attached, and the presence in the roadmap does not imply that the feature will be for sure implemented until the next release. The roadmap should guide the development of the project. Please send your suggestions, if possible with some description. As a general rule, the features to be implemented must follow the standard, if they are related directly to SIP, or they must have a clear and common usage in real cases. Cheers, Daniel Draft of the roadmap to OpenSERv1.1.0 ------------------------------------- OpenSER core: - generic communication interface which must offer an abstract layer between core/modules and transports (e.g., fifo, unix sockets) - move the code of the implemented trasports as module - NAPTR lookup - TLS multi-domain support - TLS configuration values to be set via a module, to keep core less exposed to often changes - better error reporting and handling - pseudo-variables to be introduced in different modules - security checks of destination addess (white/black lists) - memory defragmentation - tcp enhancements - OpenSER command line interface (terminal) OpenSER modules: [acc] - possibility to disable values from db_fmt [avpops] - more coherent format of the parameters (=>to be used the one from pseudo-variables) - ability to take the name of the AVP to be loaded from the value of another AVP - global avps - avps to be stored during run time, shared between processes - local avps - avps to be stored locally, specific per script, not per transaction [cpl-c] - possibility to configure table and columns' names - import registrar paramters insted of redefine [dbtext] - replace support - reload support [dispatcher] - serial forking when selected destination fails - database support [enum] [-] - parallel/serial forking based on order and preference fields [postgres] - connection pool - shift to memory manager used by openser [tm] - unification of t_relay_to_*() in a form of t_relay_to("proto:host:port") [uac] - qop authentication support - proper CSeq value after authentication challenge [usrloc] - better handling of the natted contacts, to uniquely identify a contact - cacheless usrloc - path support for registrations - sip instance support - possibility to attach to a contact a set of values (similar to log_extra in acc)

6 8

[Users] FQDN in record route
by ranveer kunal 10 Nov '05

10 Nov '05

Hi folks, I want my record route to be FQDN(domain name) instead of ip address, so i tried using record_route_preset("mysip.test"). now my problem is when i do this it also appends my name rkunal(a)mysip.test to the record route field. Is there any way of configuring openser to send FQDN in record route instead of ip address and I don't want username there. Regards, Ranveer.

2 2

[Serusers] Rewriting SDP Infos for NAT Traversal
by Anthony Chapellier 10 Nov '05

10 Nov '05

Hi, When an INVITE request is incoming, I need to rewrite VIA Header Field, Contact Header Field, (c=) field in SDP message core and (o=) field in SDP message core too with IP public adress of UAC. Those are needed by my SIP Server before SER Proxy forward the request to him. I got some success with rewriting Contact Header File by using fix_nated_contact() function and (c=) field with fix_nated_sdp("1") function. But I don't succeed at all in rewriting (o=) field and partially for VIA Header Field. To do so, I tried to use the following function : fix_nated_sdp("8"); No Result :( forcerport(); It seems to add real IP public adress and port in VIA header field but it's not rewriting IP private adress So, finally is it the good way to do this thing or is there something wrong with my behaviour ? Thanks,

2 3

Re: [Users] enable NAT OPTION ping
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

Hi Raymond, maybe the registration expires - check if when the PING stops, the registered contact is still valid (not expired). regards, bogdan Raymond Chen wrote: > Dear all, > > I want to force openser to ping all registered users with OPTIONS > message, my configuration as follow, flag in location table is 512 for > sip_nat_flag, and 513 for NAT. I can see the OPTIONS message send to > the users and got replied for a few minutes, after that, it stop > sending. Is there something I’ve done wrong? > > Thanks > > Ray > > modparam("registrar", "nat_flag", 6) > > modparam("registrar", "sip_natping_flag", 5) > > modparam("nathelper", "ping_nated_only", 0) > > modparam("nathelper", "natping_interval", 30) > > modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") > > modparam("nathelper", "sipping_method", "OPTIONS") > > modparam("nathelper", "sipping_from", "sip:keepitopen@xxxxxx.net") > > if (nat_uac_test("31")) { > > if (method == "REGISTER" || !search("^Record-Route:")) { > > fix_nated_contact(); # Rewrite contact with source IP of signalling > > if (method=="REGISTER") { > > fix_nated_register(); > > } else if (method == "INVITE") { > > fix_nated_sdp("1"); # Add direction=active to SDP > > }; > > force_rport(); # Add rport parameter to topmost Via > > append_hf("P-hint: Proxied\r\n"); > > setflag(6); # Mark as NATed > > }; > > }; > > if (uri==myself) { > > if (method=="REGISTER") { > > if (!www_authorize("", "subscriber")) { > > www_challenge("", "1"); > > exit(); > > }; > > Setflag(5); > > save("location"); > > return; > > }; > > alias_db_lookup("aliases"); > > if (lookup("location")) { > > append_hf("P-hint: usrloc applied\r\n"); > > route(1); > > return; > > }; > > }; > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >

2 1

Re: [Users] Re: [Devel] roadmap to v1.1.0
by Bogdan-Andrei Iancu 10 Nov '05

10 Nov '05

HI Raymond, I think was already mentioned in usrloc section as "cacheless usrloc" - integrating this into usrloc will be a more elegant way to do it than having another module. regards, bogdan Raymond Chen wrote: >How about usrloc-cl? > >Ray > >-----Original Message----- >From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf >Of Klaus Darilion >Sent: 2005年11月7日 18:22 >To: daniel(a)voice-system.ro >Cc: devel(a)openser.org; users(a)openser.org >Subject: [Users] Re: [Devel] roadmap to v1.1.0 > >Daniel-Constantin Mierla wrote: > > >>- generic communication interface which must offer an abstract layer >> >> >between > > >> core/modules and transports (e.g., fifo, unix sockets) >>- move the code of the implemented trasports as module >>- NAPTR lookup >> >> > >with failover to next server/protocol? interpret ICMP error messages > > > >>- TLS multi-domain support >>- TLS configuration values to be set via a module, to keep core less >> >> >exposed > >I will write a separe emails for my TLS ideas :-) > > > >>- security checks of destination addess (white/black lists) >> >> > >maybe not only based on IP address, but IP address, port and protocol >(which may be also * for all ports/protocols) > > > >>[enum] >>[-] - parallel/serial forking based on order and preference fields >> >> > >isn't this already possible using load_gw, next_gw? > > > >>[postgres] >>- connection pool >>- shift to memory manager used by openser >> >> > >there was a big patch by jan for ser - maybe we can use this patch. > >klaus > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > > > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

1 0

RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

Great! Bogdan, you are genius! I'll try it. Thank you very much! Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, November 10, 2005 17:05 To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Routing to NATed gateways if you want to try the next gateway if lookup() fails, you may try to simulate loops by recursive call of the same route block. regards, bogdan Leonid Fainshtein wrote: >Hi Bogdan, >Yes, it does. Please pay attention that the called number must be >restored in branch_route[]. >Also, I still don't know what to do if lookup() fails (the gateway is >temporary down and therefore is not registered...) Any idea? Is it >planned to add loops possibility to script language? > >Best regards, >Leonid > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Thursday, November 10, 2005 15:36 >To: Leonid Fainshtein >Cc: users(a)openser.org >Subject: Re: [Users] Routing to NATed gateways > >Hi Leonid, > >does the logic solved your problem? on a first view, the script look ok. >regarding the failure of lookup(); you can stick to sending an error >reply or you can try the next gateway provided by LCR. > >regards, >bogdan > >Leonid Fainshtein wrote: > > > >>In other words, the simplified configuration may be as the following: >> >> # Save original user name (the called phone number). >> avp_printf("$orig_called_num", "$rU"); >> if (load_gws()) >> { >> if (next_gw()) >> { >> if (lookup("location")) >> { >> avp_pushto("$ruri/username", "$orig_called_num"); >> t_on_failure("2"); >> route(x); # NAT and other usual staff there >> exit; >> } >> else >> { >> xlog("L_INFO", "I have to think what to do in this >>situation.\n"); >> sl_send_reply("404", "User Not Found"); >> } >> } >> } >> >>failure_route[2] >>{ >> if (method=="INVITE" && t_check_status("408|500|503")) >> { >> if (!next_gw()) >> { >> t_reply("503", "Service not available, no more gateways"); >> } >> else >> { >> if (!lookup("location")) >> { >> xlog("L_INFO", "I have to think what to do in this >>situation.\n"); >> t_reply("503", "Service not available, no more >>gateways"); >> exit; >> }; >> t_on_failure("2"); >> t_on_branch("2"); >> t_relay(); >> } >> } >>} >> >>branch_route[2] >>{ >> avp_pushto("$ruri/username", "$orig_called_num"); >>} >> >>Am I correct? Actually I don't know what to do if one of the gateways >>is off-line (lookup() failed). >> >>Best regards, >>Leonid Fainshtein >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Tuesday, November 08, 2005 5:30 PM >>To: Leonid Fainshtein >>Cc: users(a)openser.org >>Subject: Re: [Users] Routing to NATed gateways >> >>Hi Leonid, >> >>the scenario you are probing might be already doable. Set in the LCR >>table some IP addresses which will be used as aliases: private >>addresses in 10.10.x.x class for example. Then register this IPs in >>usrloc pointing to the real GWs address. >>So you can do lcr (next_gw) and then lookup. >> >>for this to work, lcr must operate on RURI and not on DST_URI (not >>sure >> >> > > > >>how exactly is working). >> >>regards, >>bogdan >> >>Leonid Fainshtein wrote: >> >> >> >> >> >>>Hi, >>>I'd like to use OpenSER for routing calls to NATed gateways. I also >>>want to support the LCR feature. Unfortunately, the current LCR >>>module >>> >>> > > > >>>doesn't support NAT. I mean that the gateway IP addresses must be >>>defined explicitly in the "gtw" table. Now I'm looking for a way to >>>force OpenSer to use information from the "location" table. >>>I want to write a module that will have the similar functionality >>>like >>> >>> > > > >>>the current LCR module has but resolving procedure will be more >>>complicated (not based on prefix and From only). The module should >>>build avp list where each gateway will have symbolic user name. For >>>example, g1, g2 etc. The gateways will also be registered on the >>>proxy >>> >>> > > > >>>with those names. Then I think it will be possible to call >>>lookup("location") after successful call the next_gtw() from the >>> >>> >>> >>> >>script. >> >> >> >> >>>Is it feasible? Is there another way to solve my problem? >>> >>>Thank you in advance, >>>Leonid Fainshtein >>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >>> >> >> >> >> > > > >

1 0

RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

Hi Bogdan, Yes, it does. Please pay attention that the called number must be restored in branch_route[]. Also, I still don't know what to do if lookup() fails (the gateway is temporary down and therefore is not registered...) Any idea? Is it planned to add loops possibility to script language? Best regards, Leonid -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Thursday, November 10, 2005 15:36 To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Routing to NATed gateways Hi Leonid, does the logic solved your problem? on a first view, the script look ok. regarding the failure of lookup(); you can stick to sending an error reply or you can try the next gateway provided by LCR. regards, bogdan Leonid Fainshtein wrote: >In other words, the simplified configuration may be as the following: > > # Save original user name (the called phone number). > avp_printf("$orig_called_num", "$rU"); > if (load_gws()) > { > if (next_gw()) > { > if (lookup("location")) > { > avp_pushto("$ruri/username", "$orig_called_num"); > t_on_failure("2"); > route(x); # NAT and other usual staff there > exit; > } > else > { > xlog("L_INFO", "I have to think what to do in this >situation.\n"); > sl_send_reply("404", "User Not Found"); > } > } > } > >failure_route[2] >{ > if (method=="INVITE" && t_check_status("408|500|503")) > { > if (!next_gw()) > { > t_reply("503", "Service not available, no more gateways"); > } > else > { > if (!lookup("location")) > { > xlog("L_INFO", "I have to think what to do in this >situation.\n"); > t_reply("503", "Service not available, no more >gateways"); > exit; > }; > t_on_failure("2"); > t_on_branch("2"); > t_relay(); > } > } >} > >branch_route[2] >{ > avp_pushto("$ruri/username", "$orig_called_num"); >} > >Am I correct? Actually I don't know what to do if one of the gateways >is off-line (lookup() failed). > >Best regards, >Leonid Fainshtein >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Tuesday, November 08, 2005 5:30 PM >To: Leonid Fainshtein >Cc: users(a)openser.org >Subject: Re: [Users] Routing to NATed gateways > >Hi Leonid, > >the scenario you are probing might be already doable. Set in the LCR >table some IP addresses which will be used as aliases: private >addresses in 10.10.x.x class for example. Then register this IPs in >usrloc pointing to the real GWs address. >So you can do lcr (next_gw) and then lookup. > >for this to work, lcr must operate on RURI and not on DST_URI (not sure >how exactly is working). > >regards, >bogdan > >Leonid Fainshtein wrote: > > > >>Hi, >>I'd like to use OpenSER for routing calls to NATed gateways. I also >>want to support the LCR feature. Unfortunately, the current LCR module >>doesn't support NAT. I mean that the gateway IP addresses must be >>defined explicitly in the "gtw" table. Now I'm looking for a way to >>force OpenSer to use information from the "location" table. >>I want to write a module that will have the similar functionality like >>the current LCR module has but resolving procedure will be more >>complicated (not based on prefix and From only). The module should >>build avp list where each gateway will have symbolic user name. For >>example, g1, g2 etc. The gateways will also be registered on the proxy >>with those names. Then I think it will be possible to call >>lookup("location") after successful call the next_gtw() from the >> >> >script. > > >>Is it feasible? Is there another way to solve my problem? >> >>Thank you in advance, >>Leonid Fainshtein >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > > >

2 1

[Serusers] ser authenticating with asterisk before relaying through
by Arne Van Theemsche 10 Nov '05

10 Nov '05

Hi When my SER passes through a call to asterisk, this is done without authenticating to asterisk. I use the hostname= and insecure=very method. Can't I force ser and asterisk to use authentication? thx Arne

1 0

[Serusers] Re: [Serweb-users] whats this "DB Error:no such table" error???
by Karel Kozlik 10 Nov '05

10 Nov '05

hi, > according to the "usr_preferences.sql" > (serweb/scripts/sql/usr_preferences)the attribute name and types are all > dumped into > a table called "preference_types" which is not there in the mysql DB. > this isnt a default column to be included in the tables. > ny ideas to create this table? yes ser_mysql.sh reinstall Karel > > thank you > > > ------------------------------------------------------------------------ > *From:* Karel Kozlik [mailto:karel@iptel.org] > *Sent:* Thu 10-Nov-05 2:38 PM > *To:* Divya Parvataneni > *Cc:* serweb-users(a)lists.iptel.org > *Subject:* Re: [Serweb-users] whats this "DB Error:no such table" error??? > > Hi, > that mean some table is missing in your DB. Try reinstall your db by run: > > ser_mysql.sh reinstall > > If it not help, enable logging in serweb and see the log file for > detailed info. > > Karel > > divya.parvataneni(a)wipro.com napsal(a): > > hi , > > when i try to add an attribute > > under user preferences (logged in as admin) > > it says > > > > "DB Error:no such table" > > > > can nyone pls help!! > > > > >> serweb-users(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serweb-users > > > > Confidentiality Notice > > The information contained in this electronic message and any attachments > to this message are intended > for the exclusive use of the addressee(s) and may contain confidential > or privileged information. If > you are not the intended recipient, please notify the sender at Wipro or > Mailadmin(a)wipro.com immediately > and destroy all copies of this message and any attachments. >

1 0

[Users] Please Help Me
by Ferianto monster 10 Nov '05

10 Nov '05

I am a college student in Polithecnic Caltex of Riau in Indonesia. I am in the third years now. In my last project, I want to build the VOIP with TLS system at my college because i think it is interesting. I have studied about it since one month , but i get many problems to undestand about the system. Here is some obstacles questions about the system that i have met: 1. How can the attacker eavesdrop the communication in VoIP (The way)? 2. How can TLS prevent the attacker to eavesdrop the communication? 3. How can TLS ensure the confidentially and integrity of communication? Why we don`t user other secure protocol? 4.Do all TLS` clients use Linux Operating System or they can use Windows? 5. How secure TLS to protect the communication? That is all the obstacles questions that i have met. I do hope any body can help me to answer those question. Please help me. Thank you very much. Sincerely yours, Ferianto --------------------------------- Yahoo! FareChase - Search multiple travel sites in one click.

3 2

RE: [Users] Routing to NATed gateways
by Leonid Fainshtein 10 Nov '05

10 Nov '05

In other words, the simplified configuration may be as the following: # Save original user name (the called phone number). avp_printf("$orig_called_num", "$rU"); if (load_gws()) { if (next_gw()) { if (lookup("location")) { avp_pushto("$ruri/username", "$orig_called_num"); t_on_failure("2"); route(x); # NAT and other usual staff there exit; } else { xlog("L_INFO", "I have to think what to do in this situation.\n"); sl_send_reply("404", "User Not Found"); } } } failure_route[2] { if (method=="INVITE" && t_check_status("408|500|503")) { if (!next_gw()) { t_reply("503", "Service not available, no more gateways"); } else { if (!lookup("location")) { xlog("L_INFO", "I have to think what to do in this situation.\n"); t_reply("503", "Service not available, no more gateways"); exit; }; t_on_failure("2"); t_on_branch("2"); t_relay(); } } } branch_route[2] { avp_pushto("$ruri/username", "$orig_called_num"); } Am I correct? Actually I don't know what to do if one of the gateways is off-line (lookup() failed). Best regards, Leonid Fainshtein -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Tuesday, November 08, 2005 5:30 PM To: Leonid Fainshtein Cc: users(a)openser.org Subject: Re: [Users] Routing to NATed gateways Hi Leonid, the scenario you are probing might be already doable. Set in the LCR table some IP addresses which will be used as aliases: private addresses in 10.10.x.x class for example. Then register this IPs in usrloc pointing to the real GWs address. So you can do lcr (next_gw) and then lookup. for this to work, lcr must operate on RURI and not on DST_URI (not sure how exactly is working). regards, bogdan Leonid Fainshtein wrote: >Hi, >I'd like to use OpenSER for routing calls to NATed gateways. I also >want to support the LCR feature. Unfortunately, the current LCR module >doesn't support NAT. I mean that the gateway IP addresses must be >defined explicitly in the "gtw" table. Now I'm looking for a way to >force OpenSer to use information from the "location" table. >I want to write a module that will have the similar functionality like >the current LCR module has but resolving procedure will be more >complicated (not based on prefix and From only). The module should >build avp list where each gateway will have symbolic user name. For >example, g1, g2 etc. The gateways will also be registered on the proxy >with those names. Then I think it will be possible to call >lookup("location") after successful call the next_gtw() from the script. >Is it feasible? Is there another way to solve my problem? > >Thank you in advance, >Leonid Fainshtein > >_______________________________________________ >Users mailing list >Users(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > > >

2 1

[Serusers] Important please note: unable to load mysql.so and SER Version 9.3.0: MySQL 4.1.15.
by Dhiman D Chowdhury 10 Nov '05

10 Nov '05

Dear friends, If you are new to SER like I am, please note the followings: Setup: Red Hat Linux 9.0 or Red Hat Enterprise ES 3 and SER version 0.9.3. 1. When you try to load mysql.so module for your ser, you get an error. bad config. if you do ser -D -E . The result shows that mysql.so was unable to load due to libmysqlclient.so.12 was not found. Please note that the above mentioned Red Hat Linux includes a MySQL server version 3.23 which does not include the above file. If you do prefer to upgrade your mysql server please do not upgrade it to MySQL 5.0 rather upgrade it to 4.15. You are note done yet. Go to mysql.com and try to find out the following rpm file: MySQL-shared-compat-4.1.10a-0 RPM for i386 This file is from 4.1.10 which includes libmysqlclient.so.12 that is not present 4.1.15. Once downloaded install the file using the following command: rpm -i MySQL-shared-compat-4.1.10a-0.i386.rpm --force Hope this helps. Thanks Dhiman Deb Chowdhury http://dhiman.chowdhury.com

3 2

Re: [Serusers] FW: Call logs
by Steve Blair 10 Nov '05

10 Nov '05

Colin: Ahh, I've only done work with accounting on 0.9.x systems. This might explain why my instructions didn't help. I've copied the list on this reply in the hope that someone may know the answer since I will be out of the office all day today. I'll check back with you tomorrow. -Steve Colin Jordan wrote: >Hi Steve > >Thanks for this. The SER version is 8.1. Nothing appear in the database. > >Thanks for helping, > >Colin >

1 0

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users