- sr-users - kamailio.org

[Serusers] Per-cent Sign in Attributes
by Martin Hoffmann 15 Sep '07

15 Sep '07

Salut, sorry for getting lazy now. In the config snipped | append_to_reply("%$digest_challenge"); what is the point of the per-cent sign in front of the attribute name? Thanks, Martin, who though he had figured all this out. Apparently not.

4 19

[OpenSER-Users] Problem with SRV lookups
by Tim Madorma 15 Sep '07

15 Sep '07

Hi, I have a simple application where I am trying to an SRV record to load balance towards a few different destination servers. I set up my DNS server to resolve to four different destination A record URIs - something like: _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 crunch.com. _sip._udp.aimcidfilter.com 180 IN SRV 1 50 5061 spinner.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 homer.com. _sip._udp.aimcidfilter.com 180 IN SRV 2 50 5061 bart.com. In my configuration file, I am putting a domain name in my call to seturi as follows: seturi("sip:feature_fs@aimcidfilter.com:6010"); and then I call route to send the INVITE. I was expecting to see a NAPTR and SRV record attempts, but when I sniffed DNS, I only saw A record attempts. Any advice would be very helpful. thanks, Tim ------------------------------------ My configuration file looks as follows: # # TWC OpenSER SIP Proxy for Caller ID - openser_cid.cfg # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) maxbuffer=1048576 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #reply_to_via=1 children=4 log_facility=LOG_LOCAL4 dns_use_search_list=no dns_servers_no=2 # One for the ENUM server and one for other DNS lookups user="sipproxy" group="sipproxy" disable_dns_blacklist=yes # FOR TESTING ONLY dns_try_ipv6=no # FOR TESTING ONLY # Local IP Address and port to listen to # Set the following IP addresses to the value of the local e1000g0 interface listen=udp:65.185.232.62:5060 # Alias IP address/port pair values will be in the Req URI when the # OpenSER is being load balanced with a BIG IP server # Set the IP addresses to the value of the associated blade # of the BIG IP that is load balancing for the SIP Proxy server. alias=65.185.233.104:5060 # LOAD OpenSER MODULES mpath="/sw/lib/openser/modules/" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "acc.so" loadmodule "mi_fifo.so" loadmodule "xlog.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 ------------------- modparam("acc", "log_level", 2) modparam("acc", "log_flag", 1) modparam("acc","report_cancels", 1) modparam("acc","failed_transaction_flag", 1) modparam("acc","log_extra", "req_uri=$rU") modparam("tm", "fr_timer", 5) modparam("tm","fr_inv_timer",5) modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { route(1); }; if (method=="OPTIONS") { sl_send_reply("200", "OK"); exit; } if (uri =~ "^sip:feature_fs@" || method=="NOTIFY") { xlog("L_DBG", "TWC: received incoming message:\n <$mb>\n"); seturi("sip:feature_fs@aimcidfilter.com:6010"); setflag(1); route(1); } else { sl_send_reply("404", "Not Found"); }; } ##################################################### # Default Message Handler ##################################################### route[1] { # Send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; }

3 6

[Serusers] SER Implementation
by Tuhin 15 Sep '07

15 Sep '07

Hi, I am from Bangladesh. I have downloaded and configured SER for proxy purpose. My server is working fine in test mode. Now I need some more features using SER. I wants to implement CDR and credit control module with this. My OS is Redhat Ent 4. What are the applications I need to to do so, pls let me know. Rgds Tuhin

1 0

[OpenSER-Users] Trunk Compile Error 2
by Edson 15 Sep '07

15 Sep '07

I'm trying to compile the trunk version. After updating I run make distclean and make exclude_modules="auth_diameter h350 ldap mi_xmlrpc osp postgres unixodbc" modules install but after I always get: make[1]: Leaving directory `/usr/local/src/VoIP/openser/openser/trunk/modules/auth_db' make: *** [modules] Error 2 What is this "Error 2" ? In this same machine I can compile, without a glance, branche 1.2 with this same 'make' command. Thanks for any help.... Edson

1 0

[OpenSER-Users] registration and redirection
by Mike Clark 14 Sep '07

14 Sep '07

Ok, I am brand new to openser and don't really have ti routing anything yet. But I did have a dumb or at least naive question. Can openser be configured as a very simple load balancer and redirect registration requests, in a round robin manner, to a set of Asterisk servers? I know having openser be the registrar is the best solution in the long run, but this may solve an issue quickly. I looked at the dispatch module and I know it will do want I want for invites, but didn't know if registration requests could be redirected. Thanks, Mike Clark

1 0

[OpenSER-Users] memory error sems with openser
by laurent schweizer 14 Sep '07

14 Sep '07

Hello, I user openser with sems , and each 2 week a memory error. the problem is that I don't have more trafic that usualy. Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmUtils.cpp)(write_to_socket)(614): write to unix socket: completed Sep 13 13:39:46 dns2 ./openser[3673]: t_uac: short of cell shmem Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmEventQueue.cpp)(processEvents)(71): event processed Sep 13 13:39:46 dns2 ./openser[3673]: unix_server_loop: Command 't_uac_dlg' failed with return value -1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmSession.cpp)(run)(292): 2D9B7F95-46E9218200068CF4-97601B30(a)212.203.123.82 dlg.getUACTransPending() = 1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmServer.cpp)(run)(93): revents = 1 Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmCtrlInterface.cpp)(cacheMsg)(186): recv-ed: <500 Server error occurred (2/FIFO/UAC> Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmInterfaceHandler.cpp)(handleRequest)(187): response from Ser: 500 Server error occurred (2/FIFO/UAC Sep 13 13:39:46 dns2 Sems[1254]: Error: (AmInterfaceHandler.cpp)(handleRequest)(214): reply is missing headers: <500 Server error occurred (2/FIFO/UAC> Sep 13 13:39:46 dns2 Sems[1254]: Debug: (AmServer.cpp)(run)(93): revents = 0 Sep 13 13:40:15 dns2 ./openser[3794]: ERROR:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 408 Sep 13 13:40:19 dns2 Sems[1254]: Debug: (UserTimer.cpp)(checkTimers)(97): Timeout Event could not be posted, session does not exist any more. Sep 13 13:43:19 dns2 ./openser[3764]: ERROR: new_t: out of mem: Sep 13 13:43:19 dns2 ./openser[3764]: ERROR: t_newtran: new_t failed Sep 13 13:44:23 dns2 ./openser[3748]: ERROR: new_t: out of mem: Sep 13 13:44:23 dns2 ./openser[3748]: ERROR: t_newtran: new_t failed Sep 13 13:44:28 dns2 ./openser[3781]: ERROR: new_t: out of mem: Sep 13 13:44:28 dns2 ./openser[3781]: ERROR: t_newtran: new_t failed Sep 13 13:44:47 dns2 ./openser[3765]: ERROR: new_t: out of mem: Sep 13 13:44:47 dns2 ./openser[3765]: ERROR: t_newtran: new_t failed Sep 13 13:45:09 dns2 ./openser[3747]: ERROR: new_t: out of mem: Sep 13 13:45:09 dns2 ./openser[3747]: ERROR: t_newtran: new_t failed

4 3

[Serusers] SER+Mediaproxy + X-LITE in the same LAN.
by pistara_bros＠tin.it 14 Sep '07

14 Sep '07

Hi all, i'm newbie in SER, I want to route RTP packets between 2 computers with x-lite client voip in the same network through Mediaproxy. I know this may have no sense, but it's just a test configuration. The call use_media_proxy() seems to have no effect, the RTP packets go directly from one computer to another, no packet trough Mediaproxy. This is my environment: SER and Mediaproxy on the same PC (192.168.123.186) X-lite client_A running on 192.168.123.10 X-lite client_B running on 192.168.123.168 Client_A call Client_B that accept, but mediaproxy refuses to proxy the call. This is my ser.cfg # ----------- global configuration parameters ------------------------ debug=3 # debug level listen=192.168.123.186 port=5060 log_facility=LOG_LOCAL6 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser: heslo@localhost/ser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local1/lib/ser/modules/mysql.so" loadmodule "/usr/local1/lib/ser/modules/sl.so" loadmodule "/usr/local1/lib/ser/modules/tm.so" loadmodule "/usr/local1/lib/ser/modules/rr.so" loadmodule "/usr/local1/lib/ser/modules/maxfwd.so" loadmodule "/usr/local1/lib/ser/modules/usrloc.so" loadmodule "/usr/local1/lib/ser/modules/registrar.so" loadmodule "/usr/local1/lib/ser/modules/textops.so" loadmodule "/usr/local1/lib/ser/modules/uri.so" loadmodule "/usr/local1/lib/ser/modules/uri_db.so" loadmodule "/usr/local1/lib/ser/modules/domain.so" loadmodule "/usr/local1/lib/ser/modules/auth.so" loadmodule "/usr/local1/lib/ser/modules/auth_db.so" #MEDIA PROXY module needed loadmodule "/usr/local1/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local1/lib/ser/modules/nathelper.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 2) # -- auth params -- 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") #SQL settings modparam("auth_db", "db_url", "mysql://ser: heslo@localhost/ser") modparam("usrloc", "db_url", "mysql://ser: heslo@localhost/ser") #MEDIAPROXY settings #--------------------------- # configurazione moduli #--------------------------- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy"," natping_interval", 30) modparam("mediaproxy","mediaproxy_socket"," /var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics"," /usr/local1/etc/ser/sip-clients") modparam("mediaproxy"," rtp_asymmetrics","/usr/local1/etc/ser/rtp-clients") modparam ("registrar", "nat_flag", 6) # -- 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=="INVITE" && client_nat_test("3")) { # record_route_preset("192.168.123.186:5060;nat=yes"); #} else if (method!="REGISTER") { # record_route(); #}; if (method=="INVITE" ) { if (client_nat_test("3")) { record_route_preset("192.168.123.186:5060; nat=yes"); } else record_route_preset ("192.168.123.186"); } else if (method!="REGISTER") { record_route(); }; if (method==" BYE" || method=="CANCEL") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if ((method=="INVITE" || method=="REFER") && !has_totag()) { sl_send_reply("403", "Forbidden"); break; }; if (method=="INVITE") { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply ("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; if (!uri==myself) { route(4); 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=="ACK") { route(1); break; } else if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method==" REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User 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 t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact: [ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri! =myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { if (! isflagset(8)) { setflag(8); use_media_proxy(); }; }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search ("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } And this is Mediaproxy Log: Listening for commands on local socket `/var/run/mediaproxy.sock' Listening for remote commands on `192. 168.123.186:25060' Remote commands are allowed from: 192.168.123.186, 127.0.0.1 Using IP address `192.168.123.186' for the RTP/RTCP proxy request B390Opu7cB 192.168.123.168:25500:audio 192.168.123.168 sip. antech.it local 192.168.123.186 remote unknown-agent info=from: orazio@sip.antech.it,to:j2me@sip.antech.it, fromtag:As94,totag: session B390Opu7cB: started. listening on 192.168.123.186:52010 execution time: 1.71 ms lookup B390Opu7cB 192.168.123.186:25500:audio 192.168.123.186 sip.antech.it local sip.antech.it unknown unknown-agent info=from:orazio@sip.antech.it,to:j2me@sip.antech.it,fr omtag:As94, totag:oePf execution time: 0.57 ms session B390Opu7cB: caller signed in from 192.168.123.168:25500 (RTP) (will return to 192.168.123.168: 25500) status execution time: 0.14 ms session B390Opu7cB: 2/0/0 packets, 344/0/0 bytes (caller/called/relayed) session B390Opu7cB: ended (did timeout). delete B390Opu7cB info= execution time: 0.14 ms and this is dump of session.py [root@sip mediaproxy]# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ---------------------------------------------------------------------------------------------------- 192.168.123.168:25500 - 192.168.123.186:52010 - ?.?.?.?:? inactive 0'32" GSM Audio 344/0/0 Total traffic: 0bps/0bps/0bps (in1/in2/out) Session count: 1 Proxy version: 1.9.0 Is mediaproxy working just between different networks or it's an error in my configuration file? Please help me. Any suggestions? Orazio

1 0

[Serusers] how to record conversation using rtpproxy
by nhadie 14 Sep '07

14 Sep '07

Hi, All how can i run rtpproxy to save recorded conversation locally? i tried: /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media but got nothing under /var/media after making calls /usr/local/bin/rtpproxy -l 1.2.3.4 -s udp:1.2.3.4:17000 -r /var/media -S /var/media also got nothing? has anyone tried it? thank you regards, nhadie

1 0

[OpenSER-Users] domain name in route header
by jh＠tutpro.com 14 Sep '07

14 Sep '07

klaus, if you want to support domains in route headers, then you need to do naptr and srv lookups for the domain when handling loose routing. it is not enough to compare the domain to the ones domain module knows about. domain in route header may namely be anything as long as it resolves to your proxy. for this reason, i don't like the idea of domains in route headers. it hurts performance far too much. -- juha

1 0

[OpenSER-Users] help with outbound proxy
by Dawid Jarosławski 14 Sep '07

14 Sep '07

Hello, I'm trying to make outbound proxy for my network using OpenSER. Currently I've configured OpenSER as sip registrar/rtpproxy and it works fine even with NAT traversal. The problem is that I don't how shuould I reconfigure my config to make it work as outbound proxy. I've tried many changes with no results. The first problem is how to correctly handle the register method. My config is based on: http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html Can anyone help me? Best regards, Dawid Jaroslawski

2 5

[OpenSER-Users] Patch for openserunix.c
by Patrick Miccio 14 Sep '07

14 Sep '07

Hi everyone, I use 1.1.1 notls, in my testing environment I came across some strange errors when using a home made script in combination with the openserunix util. First of all there were thousands of files like"/tmp/OpenSER.XXXXXX" but looking around with google I found a patch for that: https://sourceforge.net/tracker/?func=detail&atid=743022&aid=1708301&group_… Then with the home made script I was still getting lots of socket errors in syslog, a colleague of mine fixed the openserunix.c ############################################################ --- openserunix.c.ori 2007-09-13 19:54:00.406861237 +0200 +++ openserunix.c 2007-09-13 19:54:07.407260167 +0200 @@ -97,6 +97,7 @@ goto err; } + from_len = sizeof(from); len = recvfrom(sock, buffer, BUF_SIZE, 0, (struct sockaddr*)&from, &from_len); if (len == -1) { fprintf(stderr, "Error in recvfrom: %s\n", strerror(errno)); ############################################################ maybe this is useful for someone since even the newest OpenSER version hast the "buggy" openserunix util :-) cheers, Patrick.

4 3

[OpenSER-Users] [presence] "in-dialog" SUBSCRIBE without "Route" record
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I have presence in OpenSer same box. modparam("presence","server_address","sip:85.92.0.210") I've a problem because I get "in-dialog" SUBSCRIBE messages without "Route" header: When my SIP client (Twinkle or X-Lite) starts up it sends a SUBSCRIBE NOT-in-dialog: ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:08.294012 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:ibc@sip.domain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKcsmbires Max-Forwards: 70 To: <sip:ibc@sip.domain.net> From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 # U 2007/09/10 17:54:08.296369 85.92.0.210:5060 -> 222.21.235.18:5060 SIP/2.0 202 OK Via: SIP/2.0/UDP 192.168.1.58;rport=5060;branch=z9hG4bKcsmbires;received=222.21.235.18 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 636 SUBSCRIBE Expires: 30 Contact: <sip:85.92.0.210> Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 # U 2007/09/10 17:54:08.297746 85.92.0.210:5060 -> 222.21.235.18:5060 NOTIFY sip:angel@222.21.235.18:5060 SIP/2.0 Via: SIP/2.0/UDP 85.92.0.210;branch=z9hG4bKfaad.3f308931.0 To: sip:angel@sip.domain.net;tag=wddxn From: sip:ibc@sip.domain.net;tag=10.14184.1189439648.1 CSeq: 2 NOTIFY Call-ID: bdrmoffsihrozyj(a)192.168.1.58 Content-Length: 342 User-Agent: OpenSER (1.2.2-notls (i386/linux)) Max-Forwards: 70 Event: presence Contact: <sip:85.92.0.210> Subscription-State: active;expires=30 Content-Type: application/pidf+xml ---------------------------------------------------------------------------------------------------------- And after 60 minutes (presence refresh in X-Lite) the following SUBSCRIBE from X-Lite is this sent. Note that: - This is a IN-DIALOG SUBSCRIBE (has totag and same callerd ID as before). - This in-dialog SUBSCRIBE hasn't "Route" header !!!!!!!!!!! ---------------------------------------------------------------------------------------------------------- # U 2007/09/10 17:54:35.390759 222.21.235.18:5060 -> 85.92.0.210:5060 SUBSCRIBE sip:85.92.0.210 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.58;rport;branch=z9hG4bKpgezzksl Max-Forwards: 70 To: <sip:ibc@sip.domain.net>;tag=10.14184.1189439648.1 From: "Angel" <sip:angel@sip.domain.net>;tag=wddxn Call-ID: bdrmoffsihrozyj(a)192.168.1.58 CSeq: 638 SUBSCRIBE Contact: <sip:angel@192.168.1.58> Accept: application/pidf+xml Event: presence Expires: 30 User-Agent: Twinkle/1.1alpha2 Content-Length: 0 ---------------------------------------------------------------------------------------------------------- So, because this second SUBSCRIBE hasn't a "Route" header the funcion "loose_route()" doesn't match it and it's routed as NOT-in-dialog message. So, what to do??? I've a in-dialog message without "Route" header, so it's not routed by "loose_route()". I've tryied with this: if (loose_route() || has_totag()) { but it generates a loop. Does OpenSer store the SUBSCRIBE dialog info for 60 minutes? could it recognize this establishd dialog? or should I route this in-dialog message as NOT in-dialog? Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] About presence server in other host
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I will separate the presence server in a different host, and I'd like to know your opinions: - For SUBSCRIBE messages main OpenSer will "rewriteuri(presence_serve)" and will NOT apply "record_route()" in order to avoid in-dialog NOTIFY to pass through main OpenSer. Is it ok? - "server_address" for Presence module will have: "sip:presence-server:5060" (of course presence module will just exist in presence-server). - Any other suggestion? In which cases is necessary to do this separation? with how much SIP traffic is it interesting? Thanks for any help. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[OpenSER-Users] record_route_preset("x.x.x.x:5060; nat=yes"); and search("^Route:.*; nat=yes")
by Halomoan Chow 14 Sep '07

14 Sep '07

Dear All I don't understand why in the Openser+MediaProxy i need record_route_preset("x.x.x.x:5060;nat=yes"); and search("^Route:.*;nat=yes") What is the purpose while I can do client_nat_test to see if the request is come from a nat-ed client. If I remove search("^Route:.*;nat=yes") from below statement, I can do re-INVITE without any problem because it still have client_nat_test to find out the nat in between. if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setbflag(6); use_media_proxy(); }; Hope someone can give me a clue. Thank you. Regards, Halomoan

2 5

Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 14 Sep '07

14 Sep '07

On Thu, Sep 13, 2007 at 04:51:22PM +0200, Klaus Darilion wrote: > no - two: Right, my mistake. > Two identical NAPTR. One in RFC 3761 syntax and one in 2916 syntax. > Looks like enum_query() supports both syntax. Yes, it is my problem. I am sending this information for other users who will crawl this archive. :) > Solution 1: fix the NAPTR I sent request to their DNS administrators, it is not my site. > Solution 2: use enum_query("e164.arpa.","+sip"); (not sure about this one) I can confirm this. Works well before they update their DNS records. Thank you for help. SAL

1 0

Re[2]: [OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello Bogdan-Andrei, Friday, September 14, 2007, 10:15:55 AM, you wrote: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: method: <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: uri: <sip:820999@voip.md> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: version: <SIP/2.0> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: Found param type 235, <rport> = <n/a>; state=17 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: end of header reached, state=5 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: Via found, flags=2 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: this is the first via Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: After parse_msg... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: preparing to run routing scripts... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=100 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=9 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:820999@voip.md} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:820999@voip.md] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: to body [sip:820999@voip.md ] Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: get_hdr_body : content_length=0 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:maxfwd:is_maxfwd_present: value = 70 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: add_param: tag=gd60cu Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:parse_to:end of header reached, state=29 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBUG:parse_to: display={}, ruri={sip:checksip@sheik:37176} Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=200 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: find_first_route: No Route headers found Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: loose_route: There is no Route HF Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: rewrite_uri: Rewriting Request-URI with 'sip:820999@172.17.6.103' Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG: t_newtran: msg id=106 , global msg id=105 , T on entrance=0xffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:sl:sl_reply_error: error text is I'm terribly sorry, server error occurred (1/SL) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: parse_headers: flags=ffffffffffffffff Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]:check_via_address(Y.Y.Y.Y, sheik, 0) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DBG:sl:run_sl_callbacks: callback id 0 entered Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: ================================= Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: trace_sl_onreply_out: trace off... Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: DEBUG:destroy_avp_list: destroying list (nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11708]: receive_msg: cleaning up Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e44354 next=0xb5e3bac8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:0,tl=0xb5e3bac8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:1,tl=0xb5e3bb34 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e44344 next=0xb5e3bab8 Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11716]: DEBUG: timer routine:4,tl=0xb5e3bab8 next=(nil) Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: SIP Request: Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: method: <BYE> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: uri: <sip:X.X.X.X:5060;r2=on;lr=on;ftag=1455B87C-17C9> Sep 14 10:33:46 sip_proxy /usr/local/openser/sbin/openser[11707]: version: <SIP/2.0> > Hi, > Have you checked the openser logs to get more details about the error? > regards, > bogdan > Ion Minzu wrote: >> Hello , >> >> I have an working openser, i installed and configured in Nagios check_sip >> plugin, the problem is that openser answer to OPTIONS message with the error >> 500. >> Help me >> >> # >> U Y.Y.Y.Y:49813 -> X.X.X.X:5060 >> OPTIONS sip:820999@voip.md SIP/2.0 >> Via: SIP/2.0/UDP sheik:49813;rport >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Contact: sip:checksip@sheik:49813 >> Content-length: 0 >> Max-Forwards: 70 >> User-agent: check_sip 1.01 >> Accept: text/plain >> >> # >> U X.X.X.X:5060 -> Y.Y.Y.Y:49813 >> SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). >> Via: SIP/2.0/UDP sheik:49813;rport=49813;received=Y.Y.Y.Y. >> From: sip:checksip@sheik:49813;tag=gd60cu >> To: sip:820999@voip.md;tag=420087ba42f57437a013c6a41fc4e4c8.b0df >> Call-ID: dailgg@sheik >> CSeq: 1 OPTIONS >> Server: OpenSer (1.1.1-notls (i386/linux)). >> Content-Length: 0. >> >> >>

2 1

[OpenSER-Users] need help
by luhan 14 Sep '07

14 Sep '07

hi master-hand: i had rencontred a problem.when i configured my Openser work together with wesip,use two linux OS,one use to run openser,other run wesip, this is my configure file: openser: ------------------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) children=8 # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # port=5060 # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" "openser.cfg" 154L, 3866C 14,1 Top # 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { 61,1 49% }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(2); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } route[2]{ if(!as_relay_t("app_server_one")){ t_reply("500","Application Server error"); } 121,3-17 98% } ----------------------------- and this is wesip: ----------------------------- <Server port="8005" shutdown="SHUTDOWN" debug="0"> <Service name="Tomcat-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="0" addresses="localhost:5060" pathName="com.voztele" protocol="tcp" > <ExtraProperties> <Property key="javax.sip.SER_ADDRESS" value="192.168.7.15" /> <Property key="javax.sip.SER_PORT" value="5080" /> <Property key="STATISTICS" value="true" /> <Property key="javax.sip.STACK_NAME" value="app_server_one" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="localhost" debug="0" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> ----------------------------------------------------------------------- at the debug model: found those tips: WARNING: using only the first listen address (no fork) 0(0) init_unixsock_socket: No unix domain socket will be opened ....... bind (Cannot assign requested address) -1(0) unable to open server sockets on dispatcher -1(0) ERROR:init_mod_child: Error while initializing module seas, rank 1 -1(0) main_dontfork: init_child failed would you help me! thank you very much. All my best regards. ------------------------------ (luhan china)卢涵技术部 E-mail: luhan(a)bjybkl.com phone: 010-59828003 mobile: 13810262476 ------------------------------

1 0

[OpenSER-Users] OPTIONS problem
by Ion Minzu 14 Sep '07

14 Sep '07

Hello , I have an working openser, i installed and configured in Nagios check_sip plugin, the problem is that openser answer to OPTIONS message with the error 500. Help me # U Y.Y.Y.Y:49813 -> X.X.X.X:5060 OPTIONS sip:820999@voip.md SIP/2.0 Via: SIP/2.0/UDP sheik:49813;rport From: sip:checksip@sheik:49813;tag=gd60cu To: sip:820999@voip.md Call-ID: dailgg@sheik CSeq: 1 OPTIONS Contact: sip:checksip@sheik:49813 Content-length: 0 Max-Forwards: 70 User-agent: check_sip 1.01 Accept: text/plain # U X.X.X.X:5060 -> Y.Y.Y.Y:49813 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP sheik:49813;rport=49813;received=Y.Y.Y.Y. From: sip:checksip@sheik:49813;tag=gd60cu To: sip:820999@voip.md;tag=420087ba42f57437a013c6a41fc4e4c8.b0df Call-ID: dailgg@sheik CSeq: 1 OPTIONS Server: OpenSer (1.1.1-notls (i386/linux)). Content-Length: 0. -- Best regards, Ion Minzu, Specialist Tehnologii Informationale, I.S."Centrul de Telecomunicatii Speciale", tel:250-540, 820922 (office), 069501208 (mob), 382869185 (ICQ) mailto:ion.minzu@cts.md

2 1

[OpenSER-Users] Makefile "bug" in the latest 1.2.2 tarball...
by Geir O. Jensen 14 Sep '07

14 Sep '07

Hey, Downloaded the 1.2.2-tls tarball, extracted and did a make all - install. Everything seemed to work fine untill I tried to start it, whereupon it complained that tls was not built in. Looking at the Makefile I spotted the #TLS=1 - whence uncommented and rebuilt made everything a.o.k. I'm guessing if I did an export TLS=1 before make all it would also work? But when you give out a tarball that's supposed to include support for TLS I think I shouldn't need to do anything else than make all... Correct? So I'm calling this a bug :D // sip:gojensen@uninett.no | h323: 004710012 pstn: +47 73 55 79 23 | fax: +47 73 55 79 01

3 2

[OpenSER-Users] How to automate the new user registration process?
by lu 14 Sep '07

14 Sep '07

Hi all, Is there a simple way/opensource tool that can handle all the new VoIP users registration to the Server side automatically without human intervention/manual entry? iam using Openser 1.2 with Mysql 5.0 database. cheers.

2 1

[OpenSER-Users] Do most SIP devices support SRV DNS lookup?
by Iñaki Baz Castillo 14 Sep '07

14 Sep '07

Hi, I'd like to know about experiences using SRV DNS: - Do most SIP device support it? - Will I have unexpected surprises with some SIP phones or softphones because don't support it? - How often do the SIP devices supporting SRV have it enabled by default? Of course I will test it by myself, but I'd like to be warned about it :) Thanks for any response. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[OpenSER-Users] tls errors
by Jeremy George 14 Sep '07

14 Sep '07

Hello All, I could use some help with tls. Any pointers to a possible solution would be greatly appreciated. Please let me know if any further information would be helpful. Thanks. Everything seems to register correctly. Openserctl ul show seems correct (see below.) For a short time all is well. I can call from/to Polycom hard phones. (Eyebeam never seems to make calls, though it does register correctly.) Then calls stop working, and I begin getting the following error messages. Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_blocking_connect: SO_ERROR (111) Connection refused Sep 12 16:17:16 spongy openser[30644]: ERROR: tcpconn_connect: tcp_blocking_connect failed Sep 12 16:17:16 spongy openser[30644]: ERROR: tcp_send: connect failed Sep 12 16:17:16 spongy openser[30644]: msg_send: ERROR: tcp_send failed Sep 12 16:17:16 spongy openser[30644]: ERROR:tm:t_forward_nonack: sending request failed Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:53 spongy openser[30644]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:17:59 spongy openser[30629]: ERROR:tm:w_t_relay: t_forward_nonack failed Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:t_forward_nonack: discarding fwd for a cancelled transaction Sep 12 16:18:25 spongy openser[30625]: ERROR:tm:w_t_relay: t_forward_nonack failed I thought the errors might be caused by an underlying tcp configuration issue on the machine, so I reconfigured the proxy and uas to use tcp (but not tls) and I had no trouble. Googling and looking through the archives seems to indicate I'm not the first person to have this problem. http://www.ip-phone-forum.de/archive/index.php/t-100539.html (I don't know if there is an answer here; unfortunately, I can't read German :) Spongy% sudo openser -V version: openser 1.2.2-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, 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 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: unknown @(#) $Id: main.c 2304 2007-05-25 16:36:07Z bogdan_iancu $ main.c compiled on 11:33:05 Sep 1 Spongy% sudo openserctl ul show Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=5 max_slot=1 AOR:: didama Contact:: sip:didama@75.56.10.117:37522;transport=tls Q= Expires:: 227 Callid:: acae15de-44fdabfd-e712aa40(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: jeremy Contact:: sip:jeremy@75.56.10.115:5060;transport=udp Q= Expires:: 85 Callid:: 00036b09-50520002-40d733bc-0d83cc25(a)75.56.10.115 Cseq:: 140 User-agent:: Cisco-CP7960G/8.0 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:128.91.3.19:5060 Methods:: 4294967295 AOR:: jessica Contact:: sip:jessica@75.56.10.116:55250;transport=tls Q= Expires:: 1196 Callid:: ef97ec67-46682f92-799b9cc1(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 AOR:: peggy Contact:: sip:peggy@75.56.10.113:1210;transport=TLS;rinstance=dd48392409b03c0a Q= Expires:: 3567 Callid:: M2NhMmI1OGZkZThhYjUzMmZiZTcxY2M0MjhlODE1ZDQ. Cseq:: 4 User-agent:: eyeBeam release 1013t stamp 43069 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 5951 AOR:: helen Contact:: sip:helen@75.56.10.116:55250;transport=tls Q= Expires:: 1197 Callid:: f556d6ba-86bff9c9-f62d758c(a)75.56.10.116 Cseq:: 84 User-agent:: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 Contact:: sip:helen@75.56.10.117:37522;transport=tls Q= Expires:: 228 Callid:: dd7e9645-83cfde68-f0f8341f(a)75.56.10.117 Cseq:: 38 User-agent:: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: tls:128.91.3.19:5061 Methods:: 8063 - Jeremy

2 1

[OpenSER-Users] 513 Massage too big
by Arya 14 Sep '07

14 Sep '07

Hello everyone I have a problem connecting to OpenSER from my public IP. I can connect and make calls to OpenSER inside my LAN but when I connect from outside LAN I get the 513 Massage too big massage. I than changed the openSER default config's section that checks a message size to a bigger number and I got the I get the "483 - too many Hops" error. I changed the config back to default and tested it with ngrep and I tried connecting to the server from a server in another state. The SIP server is on 192.168.1.109 and the computer trying to connect is on 192.168.1.100 And ngrep gave me these massages: Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 67.186.60.123:29296 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 192.168.1.109:5060 -> 192.168.1.1:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.fc407ae2.0. Via: SIP/2.0/UDP 192.168.15.100:29296;received=67.186.60.123 ;branch=z9hG4bK-d87543-ce4dd823475b2c25-1--d87543-;rport=29296. To: "100"<sip:100@74.13.124.71>;tag=329cfeaa6ded039da25ff8cbb8668bd2.3724. From: "100"<sip:100@74.13.124.71>;tag=6c4a5976. Call-ID: Y2NhNzExOWI4ODViYjc2NjJlMTUxOGYwNTUxMTYxNDk.. CSeq: 1 REGISTER. Server: OpenSER (1.2.2-notls (i386/linux)). Content-Length: 0. . # U 74.13.124.71:5060 -> 192.168.1.109:5060 SIP/2.0 513 Message too big. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.3d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.2d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.1d407ae2.0. Via: SIP/2.0/UDP 192.168.1.109;received=192.168.1.1;branch= z9hG4bKcf61.0d407ae2.0. -- Thank You

4 6

Re: [OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 13 Sep '07

13 Sep '07

Hello, On Wed, Sep 12, 2007 at 05:32:15PM +0200, Klaus Darilion wrote: > 1. I have forgotten to tell you to use the "-W byline" switch too in > ngrep (much more readable SIP logs) I can resend you new logs if needed. > 2. Nevertheless, I have found the error: In the openser proxy your > create a second branch: Take a look at the logs - there are 2 INVITEs > sent from openser to ser - the only different in these INVITEs is the > branch parameter in the topmost Via header. > > The first is: > branch=z9hG4bK41a.25464943.0 > the second is: > branch=z9hG4bK41a.25464943.1 > > Thus, openser does parallel forking. Then, when Grandstream answers the > second branch (branch=z9hG4bK41a.25464943.1) openser will CANCEL the > first branch. Right. There is no problem with CANCEL but with 2 INVITEs. I have this in my openser log: 2(13970) DEBUG: t_lookup_request: no transaction found 2(13970) DBG: trans=0xb60d6e80, callback type 1, id 1 entered 2(13970) DEBUG:dialog:build_new_dlg: new dialog on hash 2386 2(13970) DBG: trans=0xb60d6e80, callback type 1, id 0 entered Is this a right place where the second fork is done? > Thus, you have either bug in your config or in your provisioning (maybe > you have the user twice in alias table or location table). Which command can create this second INVITE? What I need to find in my config? This user is not local, it is found by ENUM. My ENUM configuration: if (uri=~"^sip:\+[0-9]+@") { if (enum_query("e164.arpa.")) { xlog("L_ERR", "ENUM[e164.arpa]: [$rm] from user $fu to $ru\n"); route(5); return; }; if (enum_query("e164.arpa.", "voice")) { route(5); return; }; if (enum_query("e164.bts.sk.")) { route(5); return; }; }; Thanks. SAL

2 2

[Serusers] FW: problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum IN NS localhost IN NS e164.arpa. localhost A 127.0.0.1 e164.arpa. A 10.200.220.200 $ORIGIN 1.2.3.4.e164.arpa. IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:abcd@e164.arpa!" . $ORIGIN 8.7.6.e164.arpa. IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:xyz@e164.arpa!" . named.conf file contains, zone "e164.arpa" IN { type master; file "/var/named/named.enum"; }; This is my enum code in ser.cfg. loadmodule "/usr/local/lib/ser/modules/enum.so" modparam("enum", "domain_suffix", "e164.arpa.") if (method == "INVITE" && uri =~ "sip:\+[0-9]*+@.") { if (enum_query("e164.arpa")) { route(1); }; }; Its possible to ping the domain e164.arpa from the m/c where ser is installed. Any help in this regard is appreciated. Regards Geethanjali 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 proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] Installing OpenSER in multiprocessor machine
by John Barry 13 Sep '07

13 Sep '07

Hello, I would like to know if there are any especial considerations when compiling, configuring, and running OpenSER v1.2.2 in a multiprocessor machine. I am using a quad-processor machine (Intel Xeon @ 3.00GHz) with Red Hat Enterprise Linux AS release 4 (2.6.9-42.0.3.ELsmp #1 SMP x86_64 x86_64 x86_64 GNU/Linux). I am also using MySQL DB for authentication in the same machine. I will appreciate any advice or information. Bests, JB _________________________________________________________________ Check out some new online services at Windows Live Ideas—so new they haven’t even been officially released yet. http://ideas.live.com

2 1

[Serusers] problem in DNS lookup from enum
by geethanjali.csg＠wipro.com 13 Sep '07

13 Sep '07

Hi, I am configuring ser to use enum module. I am getting this error when dns lookup is done from ser. 0(27798) get_record: lookup(8.7.6.e164.arpa, 35) failed 0(27798) enum_query(): No NAPTR record found for 8.7.6.e164.arpa. This is my zone file: $TTL 3D @ IN SOA e164.arpa. root( 4 ; serial (d. adams) 3H ; refresh 15M ; retry 1W ; expiry 1D ) ; minimum IN NS localhost IN NS e164.arpa. localhost A 127.0.0.1 e164.arpa. A 10.200.220.200 $ORIGIN 1.2.3.4.e164.arpa. IN NAPTR 10 100 "u" "E2U+sip" "!^.*$!sip:abcd@e164.arpa!" . $ORIGIN 8.7.6.e164.arpa. IN NAPTR 20 100 "u" "E2U+sip" "!^.*$!sip:xyz@e164.arpa!" . named.conf file contains, zone "e164.arpa" IN { type master; file "/var/named/named.enum"; }; This is my enum code in ser.cfg. loadmodule "/usr/local/lib/ser/modules/enum.so" modparam("enum", "domain_suffix", "e164.arpa.") if (method == "INVITE" && uri =~ "sip:\+[0-9]*+@.") { if (enum_query("e164.arpa")) { route(1); }; }; Its possible to ping the domain e164.arpa from the m/c where ser is installed. Any help in this regard is appreciated. Regards Geethanjali 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 proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

Re: [Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Jiri Kuthan 13 Sep '07

13 Sep '07

At 11:55 12/09/2007, Frank Wang wrote: >I downloaded ser 2.0.0-rc1 source code from <http://www.iptel.org/ser_2_0_0_release_candidate_1>http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. > >When I running SER, I got follow messages: > >[root@sipmcu ser]# ../../sbin/ser -f ser.cfg >Listening on > udp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 > tcp: <http://10.0.0.21>10.0.0.21 [<http://10.0.0.21>10.0.0.21]:5060 >Aliases: > tcp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > udp: <http://sipmcu.leadtek.com.cn:5060>sipmcu.leadtek.com.cn:5060 > *: mytestser.org:* > >WARNING: no fork mode > 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) > 0(3078) Maxfwd module- initializing > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used unimportant. (I checked the code and I was a bit puzzled by it, but it really shold not matter much.) > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) WARNING: xl_mod_init: more IP <http://10.0.0.21>10.0.0.21 not used > 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] This is the problem. Maybe you have a SER instance running from the last time and you are starting the script again, for which reason SER can't be restarted -- it can't claim the same port number twice. > 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed > 0(3078) init_mod(): Error while initializing module ctl >ERROR: error while initializing modules This could have same reason. > >could anybody tell me how about it? Thanks a lot. > >Regards, >Frank >_______________________________________________ >Serusers mailing list >Serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

2 2

Re: [OpenSER-Users] WeSIP with openser 1.2.0 again
by Elias Baixas 13 Sep '07

13 Sep '07

Hi Gregorio, posting the SIP message that caused the problem would help me reproduce it, also some OpenSER logs would help. It seems to be related to the binary-encoding of the message. try out sipSpy to capture messages if you like. remove any field which you find confidential (lest Max-Forwards...) from the SIP message send it to me privately if you want. regards, Elias Gregorio wrote: > > Hi everyone, > > > > Hi! > > Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, > configuration looks alright because some SIP Servlets receive > messages. Now I am getting an exception in the SEAS module when I > receive MESSAGE, I think this problem is similar to a previous one in > the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser > 1.2.0 again]). Does someone know anything about it? > > > > This is the exception: > > > > Thanks for helping. > > > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > java.lang.ArrayIndexOutOfBoundsException: 5942 > > at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) > > at > com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) > > at > com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) > > at > com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266) > > at > com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) > > at > com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) > > at > com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) > > at > com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:357) > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException > > at > com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) > > at > com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) > > at > com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) > > at java.lang.Thread.run(Thread.java:595) > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

1 0

[OpenSER-Users] redirect server and handling 302 messages
by erik.linder＠accenture.com 12 Sep '07

12 Sep '07

Hello, I'm trying to make openser process 302 messages though I have problems(part of .cfg file below). I'll briefly explain the call flow I'd like to have: UA1 ---INVITE---> Session border controller ---INVITE--->SIP PROXY1 SIP PROXY1---INVITE--->REDIRECT SERVER---302 MOVED--->SIP PROXY1 SIP PROXY1---INVITE--->SIP PROXY2---INVITE---> Session border controller---INVITE---> UA2 This is how I want it to work. However managed to get a weird call flow were the 302 is forwarded back to the session border controller by the proxy1 which then trigger an INVITE sent to the SIP proxy2. But that is not what I want. My 302 message looks like this 20:12:34.314705 IP 192.168.200.101.sip > 192.168.200.102.sip: SIP, length: 488 E.....@.@.%....e...f........SIP/2.0 302 Moved temporarily Via: SIP/2.0/UDP 192.168.200.102;branch=z9hG4bK1658.2543f3d6.0 Via: SIP/2.0/UDP 192.168.200.10:5060;branch=z9hG4bKpis9m130dgj04b08v7k1.1 To: "22222"<sip:22222@btsip.bt.com>;tag=cc0e227ef05fff2dd833db745f3905a6.25f 2 From: "111"<sip:11111@btsip.bt.com>;tag=SDvto0501-df614a6e Call-ID: SDvto0501-515b91532ff7927a8345f39add52762f-06ahc50 CSeq: 1 INVITE Contact: , sip:22222@mydomain.com;maddr=192.168.200.103:5060;side=term> Content-Length: 0 192.168.200.101 is redirect server 192.168.200.102 is the proxy1 192.168.200.103 is proxy2 192.168.200.10 is session border controller 1. I would like to have proxy1 process the 302 message but there seems (from other mails on the mailing list) that openser does not allow routing depending on maddr in the Contact field. Sic :-( <- how can this routing decision been taken , using rewritehostport after get_redirect("*") is kind of ignored. The only workaround I have is to change the mydomain.com to the IP of proxy2 to get the automatic redirect 2. How can I block the redirect to the session border controller and send an invite right away to the proxy2? Here are the relevant part of the logic I use in proxy1 to handle the 302 I'm using this to trigger the forward to the redirect server: lookup("aliases"); #dest is remote if (!lookup("location")) { t_on_failure("3"); xlog("L_INFO","redirect invite to SLS\n"); #change orig to term in request URI $avp(s:request_uri_to_use)=$ru; avp_subst("$avp(s:request_uri_to_use)", "/orig/term/"); avp_pushto("$ru","$avp(s:request_uri_to_use)"); #forward("udp:192.168.200.101:5060"); #exit; rewritehostport("192.168.200.101:5060"); if(!t_relay()) { sl_reply_error(); } return; and this to try to handle the 302 that is received but it does not behave like I want: failure_route[3] { xlog("L_WARN","redirect FAILUREROUTE: time_t=$Ts messageid=$mi remark=here we are $bR\n"); if(t_check_status("3[0-9][0-9]")) { get_redirects("*"); xlog("L_WARN","^method=$rm ^ruri=$ru ^messageid=$mi^remark=2status is 3xx\n--$br--\nbR((($bR)))\n"); #rewritehost("$(ru{uri.maddr})"); serialize_branches(0); } xlog("L_WARN","nextbranch\n"); #next_branches(); #t_on_failure("3"); #t_relay(); return; #t_reply("404", "No branches left"); } Help would be appreciated Cheers erik This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited.

2 2

[OpenSER-Users] authorization problems with multiple servers
by Ben Smithurst 12 Sep '07

12 Sep '07

Hi We have a problem with clients authorizing (whether for INVITE or REGISTER) against a pair of openser servers, served via either round robin A records, or multiple equal preference SRV records. The problem in a nutshell is client sends INVITE to server 1 server 1 replies with authorization challenge client sends INVITE with auth to server 2 server 2 doesn't like the 'nonce' value Now, this only happens via our nat proxy, as far as we can tell, so I think the problem lies there (and I have contacted the vendor of that system). However I wonder if there is any way to make openser work in such a setup, in case other client do the same thing (which IMO is quite a stupid thing to do) ... would the valid 'nonce' values need to be stored in the DB? Incidentally we are using mysql authentication and we are using usrloc in db_mode 3, with both openser machines pointed at the same mysql server. We're using a pretty recent openser SVN revision (dev15). Any tips would be appreciated. thanks -ben -- Ben Smithurst http://www.bensmithurst.com/

2 3

[OpenSER-Users] WeSIP with openser 1.2.0 again
by Gregorio 12 Sep '07

12 Sep '07

Hi everyone, Hi! Just installed WeSIP beta-0.1.4 and openser 1.2.0 in the same machine, configuration looks alright because some SIP Servlets receive messages. Now I am getting an exception in the SEAS module when I receive MESSAGE, I think this problem is similar to a previous one in the mailing list (Sat Apr 28 16:07:34 CEST 2007, [WeSIP with openser 1.2.0 again]). Does someone know anything about it? This is the exception: Thanks for helping. java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[1]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) java.lang.ArrayIndexOutOfBoundsException: 5942 at com.voztele.seas.SeasMsg.getNboShort(SeasMsg.java:40) at com.voztele.javax.sip.header.SIPHeader.<init>(SIPHeader.java:144) at com.voztele.javax.sip.header.MaxForwards.<init>(MaxForwards.java:29) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:266 ) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPResponse.<init>(SIPResponse.java:338) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser .java:357) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595) Exception in thread "ExpressMChannel[2]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.jav a:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Thread.java:595)

1 0

[OpenSER-Users] Wanted: VoIP Engineer for Warsaw !
by laurent schweizer 12 Sep '07

12 Sep '07

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>and AVM/FRITZ!Box <http://www.fritz-shop.ch/>. Due to our rapid growth, for our new Polish subsidiary we are currently seeking for: VOIP SPECIALIST Place of work: Warsaw *Requirements:* - Graduation from college or university with a Bachelor's degree (preferably IT) - Experience with PHP - Practical knowledge of C and C++ - Practical knowledge of Mysql and Postgresql - Linux experience - Knowledge of IP Networks, UDP, TCP - Experience with tools for Network analysis like Ethereal - VoIP basic knowledge, VoIP servers, configuration of devices - Fluency in English - Ability to interact with individuals and groups at all levels - Detail oriented and analytical - Strong verbal and written communication skills Knowledge of Perl, Java, Asterisk / SER / OPENSER, ability to configure routers, Cisco or Patton gateways, knowledge of SIP and STUN protocol, knowledge of NAT problems, of outbandProxy, knowledge of monitoring tools like Cactus, Nagios, MRTG or high availability tools like DRBD, Hearthbeat would be an additional asset. Interested individuals are requested to send their resume to: ewa.ciesla(a)peoplefone.com In your application please put the following statement: I hereby agree for processing the following personal information strictly for recruitment purposes in accordance with the regulation regarding the protection of personal data passed on the following date: 29.08.97r. Dz.U nr 133 poz. 883.

1 0

[OpenSER-Users] CANCEL generated by openser after SIP/SDP Status: 200 OK
by Jan ONDREJ (SAL) 12 Sep '07

12 Sep '07

Hello, I have problems calling from my phones connected to openser to grandstream phones on another university to ser. Problem is, that my openser generates a "CANCEL" SIP packet sent to grandstream phone, but no CANCEL received from my phone. Wireshark shows something like this: 1189587144.843962 158.195.15.100 -> 158.197.16.77 SIP/SDP Status: 200 OK, with session description 1189587144.845528 158.197.16.77 -> 158.195.15.100 SIP Request: CANCEL sip:+421xxxx44903@domain.sk 1189587144.853654 158.195.15.100 -> 158.197.16.77 SIP Status: 200 canceling Communication before is OK. 158.197.16.77 is my openser server and 158.195.15.100 is peers phone. Any idea, why openser can generate this packet? Can openser generate a SIP packet not received from client? Calling from my phone to another phone on same university works well. Thank you. SAL

3 4

[Serusers] WARNING: xl_mod_init: more IP 10.0.0.21 not used and ERROR: init_tcpudp_sock: bind: Address already in use [98]
by Frank Wang 12 Sep '07

12 Sep '07

I downloaded ser 2.0.0-rc1 source code from http://www.iptel.org/ser_2_0_0_release_candidate_1 and install on my Linux step by step according the tutorial on the WEB. When I running SER, I got follow messages: [root@sipmcu ser]# ../../sbin/ser -f ser.cfg Listening on udp: 10.0.0.21 [10.0.0.21]:5060 tcp: 10.0.0.21 [10.0.0.21]:5060 Aliases: tcp: sipmcu.leadtek.com.cn:5060 udp: sipmcu.leadtek.com.cn:5060 *: mytestser.org:* WARNING: no fork mode 0(3078) init_tcp: using epoll_lt as the io watch method (auto detected) 0(3078) Maxfwd module- initializing 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) WARNING: xl_mod_init: more IP 10.0.0.21 not used 0(3078) ERROR: init_tcpudp_sock: bind: Address already in use [98] 0(3078) ERROR: ctl: mod_init: init ctrl. sockets failed 0(3078) init_mod(): Error while initializing module ctl ERROR: error while initializing modules could anybody tell me how about it? Thanks a lot. Regards, Frank

1 0

Fwd: [OpenSER-Users] Transaction Deletion
by Brandon Armstead 12 Sep '07

12 Sep '07

---------- Forwarded message ---------- From: Brandon Armstead <brandon.armstead(a)gmail.com> Date: Sep 7, 2007 12:06 PM Subject: Re: [OpenSER-Users] Transaction Deletion To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Hello Bodgan, There is no noticable delay however the only thing I see is that one of the RR'ed proxy's responds sends out 200 OK rather than the original. You can see an example sipcap of this at this url: http://pastebin.ca/685897 . If you could let me know if you see anything that looks rather, odd, towards the end you can see the last few transactions and where there is no response to 216.82.224.203's 200 OK, where it flips from 216.82.224.202 to 216.82.224.203 from the RR. Also then later the PSTN follows with a BYE, due to no response, once again thanks for your help! Regards, Brandon On 9/7/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > Hi Brandon, > > Is there any noticeable delay between the 200OK and ACK ? > > regards, > bogdan > > Brandon Armstead wrote: > > Hello Users, > > > > I have a weird scenario where the PSTN is responding with 200 OK, > > however Openser Never forwards on the ACK, after several > > retransmissions of 200OK to Openser, PSTN responds with BYE, anyone > > know what might be going on? I've played with the timers to no avail, > > any help is appreciated, thank you! The log below is as follows: > > > > 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 > > 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 > > 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : removing 0x2b78e4e73338 from table > > 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > delete transaction 0x2b78e4e73338 > > 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: > > wait_handler : done > > 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:parse_to:end of header reached, state=10 > > 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_field: <To> [26]; uri=[sip:openser] > > 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to > > body [<sip:openser> ] > > 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: > > get_hdr_body : content_length=0 > > 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: > > DEBUG:destroy_avp_list: destroying list (nil) > > 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:forward_reply: found module tm, passing reply to it > > 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: start=0xffffffffffffffff > > 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > add_param: tag=gK0288a919 > > 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: > > DEBUG:parse_to:end of header reached, state=29 > > 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > get_hdr_field: <To> [53]; uri=[ sip:17144828306@openser] > > 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to > > body [<sip:17144828306@openser>] > > 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: hash 53903 label 1526431020 branch 0 > > 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: no matching transaction exists > > 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_reply_matching: failure to match a transaction > > 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: > > t_check: end=(nil) > > > > It seems like its not matching the transaction, because a timer is > > removing the transaction? Any ideas? Thanks! > > -- > > Brandon Armstead > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > -- Brandon Armstead -- Brandon Armstead

2 1

Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 12 Sep '07

12 Sep '07

Arya, it means you have a routing error in the script - you need to debug your script to see where the routing logic is broken - try using xlog() messages to see the execution path of your script. I cannot help you with more than this. regards, bogdan Arya wrote: > any idea how I can fix the problem? > > On 9/11/07, *Arya* <arya6000(a)gmail.com <mailto:arya6000@gmail.com>> > wrote: > > I used ngrep but it just shows that the massage keeps bouncing > server to client, and eventually it says massage too big > > > On 9/11/07, * Bogdan-Andrei Iancu* < bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > use ngrep or tcpdump to get a network trace from the proxy > server, on > the loopback interface. > > Regards, > Bogdan > > Arya wrote: > > Can you tell me how I can trace it? could this be an issue > with my > > router? its the Linksys Wr54G and its set on DMZ to the server > > > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro> > > <mailto: bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>> wrote: > > > > Hi Arya, > > > > I suspect you have a routing problem that cause the > request to loop on > > openser (via loopback interface) - openser keeps sending the > > request to > > itself. That's why you get a "Message to big" or "too > many hops".... > > > > run a trace on lo to see if this is what's happening. > > > > regards, > > bogdan > > > > Arya wrote: > > > Hello everyone > > > > > > I have a problem connecting to OpenSER from my public IP. > > > > > > I can connect and make calls to OpenSER inside my LAN > but when I > > > connect from outside LAN I get the 513 Massage too big > massage. > > I than > > > changed the openSER default config's section that > checks a message > > > size to a bigger number and I got the I get the "483 - > too many > > Hops" > > > error. > > > > > > I changed the config back to default and tested it with > ngrep and I > > > tried connecting to the server from a server in another > state. > > > > > > The SIP server is on 192.168.1.109 > <http://192.168.1.109> <http://192.168.1.109 > <http://192.168.1.109>> > > <http://192.168.1.109> and the > > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> > > < http://192.168.1.100> <http://192.168.1.100 < > http://192.168.1.100>> > > > > > > And ngrep gave me these massages: >

1 0

[OpenSER-Users] 'while' statement in config file
by Daniel-Constantin Mierla 12 Sep '07

12 Sep '07

Hello, although is more part of devel, the decision will affect users of future stable releases, so I posted on both lists. Maybe some of you noticed that in the ongoing process of improving the pseudo-variables system, the AVPs and HDRs can be accessed with dynamic index. That means next lines are correct in the config file. $var(i) = 1; xlog("$(avp(i:20)[$var(i)])"); if($(avp(i:20)[$var(i)]) == 3) { } In the past, the index was static, as a number. The question I want to address is whether makes sense to add 'while' statements to iterate through avps or hdrs. More or less, the cycles can be done via recursive routes. The block: $var(i) = 0; while($(avp(i:20)[$var(i)]) != null) { ..... $var(i) = $var(i) +1; } is equivalent with: $var(i) = 0; route(1); .... route[1] { if(($(avp(i:20)[$var(i)]) == null) return; ... $var(i) = $var(i) +1; route(1); } 'while' may look more attractive, but could open the pandora box, people asking for more and more, that will turn config language is something very complex (not saying that is simple now :-) ). The drawback with recursive routes is the fixed number of routes that can be defined with the standard openser: 40 (number can be increased if openser is recompiled), because for each cycle another route has to be used. Waiting for your opinions... Cheers, Daniel

3 2

[OpenSER-Users] add/remove headers and append_branch
by Madalina Pitis 12 Sep '07

12 Sep '07

Hello, I have a problem trying to remove a header using remove_hf() after an append_branch(). I am using 1.2 version of OpenSer. The scenario with the problem is as described in the folowing lines. When I receive an INVITE I try a parallel fork (one leg towards PSTN, the other towards IP), at which time I append a "Redirection" header. The PSTN leg will be answered with a 486 response, due to the presence of the "Redirection" header. When the IP leg expires, I receive a 408 response, at which time I do a serial fork, and resend the message towards the PSTN destination, this time without the "Redirection" header. The problem is that I cannot seem to be able to remove (or find) the header. I tried everywhere (before appending the new branch, after, in branch_route, before relaying), as you will see below. I tried searching the mailing lists but nothing turned up. I would greatly appreciate any help, for I am running out of ideas. Here is a simplified version of my configuration script, containing only the significant bits of the routing blocks: #--------------------------------------------------- # main routing logic #--------------------------------------------------- route { # check maxfwd and message lenght # do loose routing if (method=="INVITE") { # rewrite ruri rewriteuri(<PSTN destination>); # signal forking append_hf("Redirection: count=5\r\n"); append_branch(); revert_uri(); t_on_failure("1"); if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } else if (method=="ACK" || method=="BYE" || method=="CANCEL") { # default message handler } else { # forbidden message handler } } #--------------------------------------------------- # Failure route relay #--------------------------------------------------- route[1] { xlog("acc_info", " Failure route relay"); # try removing it before t_relay if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before t_relay"); remove_hf("Redirection"); } if (!t_relay()){ xlog("acc_info", " ERROR"); sl_reply_error(); } } #--------------------------------------------------- # Failure route handling #--------------------------------------------------- failure_route[1] { xlog("acc_info", " failure route handler"); if(t_check_status("408")) { t_on_branch("1"); rewriteuri(<PSTN destination>); # try removing it before append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove before append branch"); remove_hf("Redirection"); } append_branch(); # try removing it after append branch if(is_present_hf("Redirection")){ xlog("acc_info", " Remove after append branch"); remove_hf("Redirection"); } route(1); } }

2 2

[OpenSER-Users] append_hf() and failure_route issue
by Alex Hermann 11 Sep '07

11 Sep '07

Hello all, I'm having some trouble with failure routes and custom SIP headers. In FAILURE_ROUTE I set the ruri to a new destination. This destination needs a custom SIP header. When I add a SIP header with append_hf() (from within the failure_route), this header is correctly sent to the first IP of the destination on t_relay(). However, when this destination fails with 503 or timeout, the built-in DNS based failover doesn't add the custom SIP header to the new requests for subsequent IP's. When the SIP header is added from REQUEST_ROUTE the header is sent to all destinations by the DNS based failover mechanism. How do I get the DNS based failover to send my SIP headers all the time? (The same problem is present with uac_auth()) -- Greetings, Alex Hermann

1 0

Re: [OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by Iñaki Baz Castillo 11 Sep '07

11 Sep '07

First of all sorry for this post not being in the thread it belongs to (I've not the initial mail). > But, I don't want to use the Asterisk dtmf features, > I want to use the SIP INVITE/REFER features from the > UAC (actually polycom and snom phones). > > Can someboy shed me some light on this? You receive a call from Asterisk UAC and want to transfer (blind or attender) to other UACregistered in OpenSer. Of course you don't need Asterisk native transfer, SIP allows blind and attended transfer. The only you need is that your SIP devices allow both of them (unfortunatelly many SIP devices just allow blind transfer, but in most cases reading the device manual you'll find how to do a attended transfer). The point here is that Asterisk allows receiving a REFER, so don't worry for that (I use your same escenary with no problem and SIP transfers). Maybe you find useful this links: http://tech-invite.com/Ti-sip-service-4.html http://tech-invite.com/Ti-sip-service-5.html http://dev.openwengo.com/trac/openwengo/trac.cgi/wiki/SipTransfer Regards. -- Iñaki Baz Castillo

1 0

Re: [OpenSER-Users] 513 Massage too big
by Bogdan-Andrei Iancu 11 Sep '07

11 Sep '07

Hi, use ngrep or tcpdump to get a network trace from the proxy server, on the loopback interface. Regards, Bogdan Arya wrote: > Can you tell me how I can trace it? could this be an issue with my > router? its the Linksys Wr54G and its set on DMZ to the server > > On 9/10/07, *Bogdan-Andrei Iancu * <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>> wrote: > > Hi Arya, > > I suspect you have a routing problem that cause the request to loop on > openser (via loopback interface) - openser keeps sending the > request to > itself. That's why you get a "Message to big" or "too many hops".... > > run a trace on lo to see if this is what's happening. > > regards, > bogdan > > Arya wrote: > > Hello everyone > > > > I have a problem connecting to OpenSER from my public IP. > > > > I can connect and make calls to OpenSER inside my LAN but when I > > connect from outside LAN I get the 513 Massage too big massage. > I than > > changed the openSER default config's section that checks a message > > size to a bigger number and I got the I get the "483 - too many > Hops" > > error. > > > > I changed the config back to default and tested it with ngrep and I > > tried connecting to the server from a server in another state. > > > > The SIP server is on 192.168.1.109 <http://192.168.1.109> > <http://192.168.1.109> and the > > computer trying to connect is on 192.168.1.100 > <http://192.168.1.100> <http://192.168.1.100 <http://192.168.1.100>> > > > > And ngrep gave me these massages: > > > > > > > > > -- > Thank You

1 0

[OpenSER-Users] Openser and Voice master
by Dan 11 Sep '07

11 Sep '07

Hi, Has anybody able to successfully able to route calls to voicemaster from openser. I tried all method but none worked. From username/password to IP based authentication to prefix based authentication. Regards Dan

1 0

[OpenSER-Users] OpenSER+Asterisk: Attended Transfer, Call pickup
by perl＠libkvm.org 11 Sep '07

11 Sep '07

Hello, I am configuring a platform unig OpenSER 1.2 with Asterisk 1.2. I have all the SIP phones registered on OpenSER. I use the realtime setup on Asterisk using a view on MySQL. I need to be able to process the attended transfer between two UAC registered on OpenSER, having the first received a call from the Asterisk side, so that the second UAC gets the call transferred. I also need to do this as a blind transfer. But, I don't want to use the Asterisk dtmf features, I want to use the SIP INVITE/REFER features from the UAC (actually polycom and snom phones). Can someboy shed me some light on this? I need to evaluate if this is possible, or if it needs lots of sources modifications. Thank you very much.

1 0

[Serusers] ERROR: warning_builder: buffer size exceeded
by Kostas Marneris 10 Sep '07

10 Sep '07

Hello all, I have occasionally the following warning mesgs at my SER log : Sep 10 14:34:27 serhost ser[17957]: ERROR: warning_builder: buffer size exceeded Sep 10 14:34:27 serhost ser[17957]: WARNING: warning skipped -- too big I noticed that the R-URI in that case is : R-URI=<sip:XXXXXXXXXX@mydomain.com;user=phone;transport=UDP> when usually the R-URI is : <sip:XXXXXXXXXX@mydomain.com;transport=UDP> <sip:XXXXXXXXXXXXX@mydomain.com> Anyone seen those mesgs before ? Can this explained by the extra field 'user=phone' which the SIP UA adds at the R-URI of the INVITE mesg ? thanks Kostas

3 2

[OpenSER-Users] Problem with rtpproxy
by Jeferson Prevedello 10 Sep '07

10 Sep '07

Hi Guys, When I originate a call for an user behind of a NAT the call is without audio in the two ways. I believe that the problem is related to rtpproxy, however did not find nothing of wrong in my configuration. Some tip? openser.cfg: # ------------- !! Nathelper modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # Nathelper with RTPproxy service run: ps -aux | grep rtp root 26098 0.0 0.0 2532 428 ? Ss 10:40 0:00 /usr/bin/rtpproxy root 11143 0.0 0.0 2536 352 ? Ss 11:30 0:00 /usr/bin/rtpproxy -l xxx.xxx.xxx.xxx -s udp:xxx.xxx.xxx.xxx 7890 logs: Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11859]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11860]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11861]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Sep 10 11:32:59 src@sip2 /usr/sbin/openser[11862]: rtpp_test: RTP proxy <unix:/var/run/rtpproxy.sock> found, support for it enabled Regards Jeferson

2 2

[Serusers] Need SIP proxy to translate UDP<->TCP
by Thermal Wetland 10 Sep '07

10 Sep '07

Hello, I would like to hire someone to setup a SIP UDP<->TCP proxy using SER. SER will sit between an Asterisk box and Microsoft Exchange 2007. Asterisk speaks UDP & the Exchange box speaks TCP. I will make a fresh fully patched CentOS 5 box available for the work. Payment could be through PayPal or Rent a Coder or similar. I would expect an SER expert to be able to do this in less than 2 hours. Please let me know if you are interested - thermalwetland at gmail dot com. Thanks, Thermal

2 1

Re: [OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Hi Bogdan I got it fixed. The problem is on this line: if(!t_relay("0x05")) if I change to if(!t_relay()) everything is ok. Don't know what is 0x05 mean. Regards, Halomoan On 9/10/07, Halomoan Chow <halomoan(a)gmail.com> wrote: > > Hi > > My Openser version is openser-1.2.0-notls and the Mediaproxy is > mediaproxy-1.9.0 > I was using the openser.cfg that was generated from > http://www.sipwise.com/wizard/ for Openser + Mediaproxy only. > > Just only simple call from A to B cause segmentation fault. > > I don't know how to locate the core file and get the backtrace :( > > Regards, > > Halomoan > > On 9/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > > > > Hi Halomoan, > > > > provide full details about your openser version - version, how did you > > get it (source, binaries, etc). > > also locate the core file and get the backtrace. > > > > Regards, > > Bogdan > > > > Halomoan Chow wrote: > > > Dear All > > > > > > I'm running into segmentation fault error on Openser 1.2.x with > > > Mediaproxy. > > > What is the first thing I should take a look into to find out which > > > line or which module causing this segmentation fault? > > > Thank you. > > > > > > Regards, > > > > > > Halomoan > > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

3 3

Re: [Serusers] [Serdev] Release of SER 2.0 (and future 2.1)
by Greger V. Teigre 10 Sep '07

10 Sep '07

Ok, I conclude then that final release of SER 2.0 is aimed at October 15. That should allow us to align SER, serctl, SERWeb, rtpproxy, and SEMS. SER, SERWeb, and SEMS will have independent releases, and they will be aligned to be interoperable. SER, rtpproxy, and serctl will have a common release. The release will include the new buildsystem that was tested in release candidate 1. The config file and features available work with sercmd, serctl, as well as with SERWeb. SEMS support is planned, but pending some patches. All components will be combined in a one-file download with sources for all components, the SIP Express Bundle. I urge everybody with patches, bug reports, or tasks related to this release to prioritize those now! g-) Greger V. Teigre wrote: > Guys, > My last post concerned documentation. I suggested decoupling > documentation from the CVS to allow us to work independently with code > and docs. > > I suggest we move forward on the decoupling and set a date for SER 2.0 > release. My suggestion is that we target October 15. This will allow > people to allocate time to do the last clean-ups, bug-fixes, and patches. > > I also suggest that we set a release date for 2.1. AFAIU, there is > already quite a lot of code that has gone into the trunk and as 2.1 is a > minor release and we strive for ser.cfg compatibility with 2.0, we > shouldn't really wait too long. > > I suggest mid-January 2008. How does that work for you? > g-) > _______________________________________________ > Serdev mailing list > Serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev > > >

1 0

[OpenSER-Users] Openser 1.2.x Segmentation Fault
by Halomoan Chow 10 Sep '07

10 Sep '07

Dear All I'm running into segmentation fault error on Openser 1.2.x with Mediaproxy. What is the first thing I should take a look into to find out which line or which module causing this segmentation fault? Thank you. Regards, Halomoan

2 1

[OpenSER-Users] How do I obtain or log the return code in a failure_route[]?
by Juan Carlos Castro y Castro 10 Sep '07

10 Sep '07

I know I can test the SIP return code with an "if", but how do I log it, or, alternatively, pass it to a exec_dset() script? I tried something like log(1, "Failure Route, status is $status\n"); And I get a "bad logargument" when I start openser. Clues? Juan

4 6

[OpenSER-Users] Openser Signaling and Routing
by Kaeman Chris 08 Sep '07

08 Sep '07

Hello, Is this possible? Call comes to openser for handling signaling, and is handed over to B2BUA for topology hiding and media handling. But the routing(selection of gw to terminate to the supplier) is used from openser LCR. Or after the call is given to B2BUA the routing is to be performed by the B2BUA. Thanks!! Regards, KChris --------------------------------- 5, 50, 500, 5000 - Store N number of mails in your inbox. Click here.

2 1

[OpenSER-Users] Empty $du
by Andreas Sikkema 07 Sep '07

07 Sep '07

Hi, I'm trying to save the destination IP address of an INVITE message received and processed by OpenSER in the acc table. I've added an extra column and told the acc module about the new column. When I looked at the contents of the new column it was empty and adding an xlog() call to the config file also showed an empty $du pseudo variable. I know the OpenSER version I'm using is not the very latest, but it should be new enough to properly fill $du. I'm calling acc for received INVITE and BYE messages and showing $du with xlog() in the last line of INVITE processing for my testcase when the destination should be available. What am I doing wrong? -- Andreas Sikkema

2 1

[OpenSER-Users] modparam("usrloc", "nat_bflag", 6) and setflag(6) doesn't update flags column
by Halomoan Chow 07 Sep '07

07 Sep '07

Dear all I followed tutorial from SER-Getting Start document for Mediaproxy to solve nat problem. I read on the Openser document http://www.openser.org/docs/modules/1.2.x/usrloc.html#AEN83 I have to use modparam("usrloc", "nat_bflag", 6) instead of modparam("registrar", "nat_flag", 6) And from the documentation, it said "When SIP clients attempt to REGISTER with our SIP proxy we need a way to tell the registrar module to store NAT information for any particular UA. We do this by using flag 6, which has been arbitrarily chosen (but defined earlier in the loadmodule parameter section). We could have specified another integer here, but flag 6 seems to be the accepted standard for nat_flag. If the nat_flag is set before calling the save() function to store contact information, SER will preserve the NAT contact information as well as set the flags column in the MySQL location table. By doing so, we can call lookup(location) when processing messages and flag 6 will be set for NATed clients." I have a NAT UA is detected as NAT-ed UA by Openser with below rule and do setflag(6); before it does save("location"); Why in mysql, location table, i see the flags column still 0 instead of 6 ? Did I miss something? Please help :( # $Id: nat-mediaproxy.cfg 51 2006-01-31 13:28:04Z /CN=Paul Hazlett/emailAddress= paul(a)onsip.org $ debug=3 fork=no log_stderror=yes #alias="halodns.powere2e.com" listen=x.x.x.x # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no 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_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/mi_fifo.so" modparam("mi_fifo","fifo_name","/tmp/openser_fifo") modparam("auth_db|domain|uri_db|usrloc", "db_url", " mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper|registrar", "received_avp", "$avp(i:42)") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) <- set to 6 modparam("rr", "enable_full_lr", 1) . . . . route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_NOTICE","$Cxg Enter Route[2]: REGISTER $Cxx\n"); sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("2")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; }

2 1

[OpenSER-Users] uac_auth, how to sending request in failure route?
by Cseke Tamas 07 Sep '07

07 Sep '07

Hello, I 'd like to relay calls to another sip proxy, which requires authentication. But i can't forward the 407 reply to the caller, beacause they cant authenticate themselves, so i have to authenticate the call myself, without the caller knows anything about it. witch uac_auth, i can add the proxy-authorization header, but i can't send the new request to the proxy. i try this way: route[1] { t_on_failure("1"); rewritehostport("another.proxy:5060"); if (!t_relay()) { sl_reply_error(); } exit; } failure_route[1] { xlog("L_WARN", "FAILURE Route 3 AUTH required...trying..\n"); uac_auth(); if(t_newtran()) { xlog("L_WARN", "Creating new transaction successfully...\n"); } else { xlog("L_WARN", "Creating new transaction failed...\n"); }; t_relay(); } When the proxy challange authentication with 407, the call is passed to failure route properly, but t_relay failed with the following error: 0(27975) ERROR:tm:t_forward_nonack: no branch for forwarding 0(27975) ERROR:tm:w_t_relay: t_forward_nonack failed I think that's mean that the transaction doesn't exists, becacause it acknowleged alrealy, am i right? i think i have to t_newtran(), but it cannot be done in failure route, Can you suggest any method to send the request again with the credentials? Thanks any help, Tamas

2 1

[OpenSER-Users] my small security breach REGISTER
by Marc LEURENT 07 Sep '07

07 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a security matter with my configuration (default one), it's possible to register using login/password and to set anything in the contact field. So if you have an account 106/password, it's possible to be 105 in the location database! How is it possible to deny that kind of matter..? Thanks Is it useful to use: method_filtering of the REGISTRAR module Or is it better to so something whith the values below and a compare function?? $ct - reference to body of contact header $ar - realm from Authorization or Proxy-Authorization header $au - username from Authorization or Proxy-Authorization header if ($ct != $au@$ar) { sl_send_reply("403", "User and login must be the same"); }; Best Regards, Marc LEURENT # U 82.127.0.79:1045 -> 88.191.45.91:5060 REGISTER sip:sd-7501.dedibox.fr;user=phone SIP/2.0. Via: SIP/2.0/UDP 82.127.0.79:1046;branch=z9hG4bK5808036470869310420. From: <sip:105@sd-7501.dedibox.fr:5060;user=phone>;tag=c0a80101-38c0e7. To: <sip:105@sd-7501.dedibox.fr:5060;user=phone>. Call-ID: 29eb6e9-c0a80101-5-17(a)192.168.95.70. CSeq: 90 REGISTER. Max-Forwards: 70. Expires: 3600. Contact: <sip:105@82.127.0.79:1046;user=phone>. Authorization: Digest username="106", realm="sd-7501.dedibox.fr", nonce="46dfceb402cad04812873b855bc50ea65aa99ed5", uri="sip:sd-7501.dedibox.fr", response="7dca83fd358a9aea3a963f4a71ea5c9e", algorithm=MD5, qop=auth, cnonce="38c102", nc=00000001. User-Agent: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4. Allow-Events: refer,dialog,message-summary,check-sync,talk,hold. Content-Length: 0. . AOR:: 105 Contact:: sip:105@82.127.0.79:1046;user=phone Q= Expires:: 194 Callid:: 29eb6e9-c0a80101-5-17(a)192.168.95.70 Cseq:: 92 User-agent:: THOMSON ST2030 hw0 fw1.56 00-0E-50-4E-AF-C4 Received:: sip:82.127.0.79:1045 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:88.191.45.91:5060 Methods:: 4294967295 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG39AIqjpLE0HiOBYRAiUKAJ9Ilv+Zpbzw89tqWgwmHyVjU/DXugCgjEh8 5XQKEAeiF/L4RWszGC2/yzQ= =SXE9 -----END PGP SIGNATURE-----

6 7

[OpenSER-Users] is_uri_host_local() returns 1 or 4294967295 (always TRUE !!!)
by Iñaki Baz Castillo 07 Sep '07

07 Sep '07

Hi, in order to minimize the SQL queries for functions "is_uri_host_local()" and "is_from_local()" I want to exec they just one time and store the result in a variable, but the result is unexpected: $var(is_uri_host_local) = is_uri_host_local(); xlog("------------is_uri_host_local = $var(is_uri_host_local)\n"); So if the URI is local then the result is 1. But if the URI is not local then result is 4294967295. ¿?¿?¿?¿ Obviously it makes me difficult for me to do: if ($var(is_uri_host_local)) because always is true and I need to convert the values and so. Why "is_uri_host_local()" doesn't return -1 in case of being not local domain? Or maybe the issue is when I store the result in a variable? but why? Thanks. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 10

[OpenSER-Users] Transaction Deletion
by Brandon Armstead 07 Sep '07

07 Sep '07

Hello Users, I have a weird scenario where the PSTN is responding with 200 OK, however Openser Never forwards on the ACK, after several retransmissions of 200OK to Openser, PSTN responds with BYE, anyone know what might be going on? I've played with the timers to no avail, any help is appreciated, thank you! The log below is as follows: 375 Sep 6 12:18:10 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:1,tl=0x2b78e4e73588 next=(nil), timeout=261 376 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: timer routine:2,tl=0x2b78e4e733b8 next=(nil), timeout=262 377 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : removing 0x2b78e4e73338 from table 378 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: delete transaction 0x2b78e4e73338 379 Sep 6 12:18:11 openser /usr/local/sbin/openser[17385]: DEBUG: wait_handler : done 380 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:parse_to:end of header reached, state=10 381 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_field: <To> [26]; uri=[sip:openser] 382 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: to body [<sip:openser> ] 383 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG: get_hdr_body : content_length=0 384 Sep 6 12:18:12 openser /usr/local/sbin/openser[17345]: DEBUG:destroy_avp_list: destroying list (nil) 385 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:forward_reply: found module tm, passing reply to it 386 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: start=0xffffffffffffffff 387 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: add_param: tag=gK0288a919 388 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG:parse_to:end of header reached, state=29 389 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: get_hdr_field: <To> [53]; uri=[sip:17144828306@openser] 390 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: to body [<sip:17144828306@openser>] 391 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: hash 53903 label 1526431020 branch 0 392 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: no matching transaction exists 393 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_reply_matching: failure to match a transaction 394 Sep 6 12:18:14 openser /usr/local/sbin/openser[17343]: DEBUG: t_check: end=(nil) It seems like its not matching the transaction, because a timer is removing the transaction? Any ideas? Thanks! -- Brandon Armstead

2 1

[OpenSER-Users] Dispatcher + NAT
by kokoska rokoska 07 Sep '07

07 Sep '07

Hi all, I'm currently succesfully running OpenSER, but user population grow up and I'm affraid I need laod balancing :-) The projected scenario is: USERS <-> (NAT) <-> LB <-> P1 <-> GWs <-> P2 <-> R1 <-> R2 Where LB is Load Balancer (OpenSER with Dispatcher & Path module), and P1-P2 is proxies (OpenSER) and R1-R2 registrars (OpenSER) And the problem I need to solve is NAT handling. I can correctly process registrations with help of Path module on LB and enabling of using it on R1,R2. But, what should I do with calls? My idea is to process NAT handling on P1-P2, but I don't know how. Both proxies are unable (I mean) detect correct user IP address. Could you tell me, please, what to do? Do NAT handling on LB? It is unnecessary added load to LB... Thanx in advance! kokoska.rokoska

2 3

[OpenSER-Users] is_from_user_enum() errors
by Eric Hiller 07 Sep '07

07 Sep '07

Hello all, I have a rather strange issue, I cannot use "is_from_user_enum()" without first using the variable $fu somewhere; or else I get the error "0(5858) get_parsed_from(): From header is not properly parsed". Originally I had an xlog there and it all worked fine, but when I took it away I started getting the error. Now I just have " $var(bleh) = $fu;" there and it makes it work fine. I had a forum post @ http://www.voipuser.org/forum_topic_10875.html that shows my entire config. Thanks for any help, Eric _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

2 1

[OpenSER-Users] CDRTool Price is all 0.0000
by Chahn Kim 06 Sep '07

06 Sep '07

Hi all. CDRTool shows all call records with price o.oooo. Syslog logs following: Error: Cannot find rates for callid=, domain=koreanet.us, destination=1847, profile=441, app=audio Can anyone help? Thanks in advance John Chahn Kim email: johnk(a)koreanet.us

1 0

[OpenSER-Users] Limiting maximum simultaneous calls per user
by Sebastian Mangelkramer 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear list, actually I'm searching for a solution to limit the max. simultaneous calls per user. I`ve searched the internet and list-archives for possible solutions. But until now i haven`t found a solution. What i exactly need is to define the max. simultaneous calls per user (e.g. 2,4,6,8,...). I know, that openSER is transaction-stateful, not call-stateful but is there a way to monitor the number of RTP-Sessions per user from Mediaproxy and set the "max_calls"-variable via an AVP? My setup is openSER 1.2.1-notls + Mediaproxy Has anyone implemented such a solution or anyone know such one? Perhaps i had to place a kind of SBC/B2BUA in front of my SIP-Proxy? Many thanks in advance, kind regards Sebastian -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (Darwin) iD8DBQFG3tU87EKQHe5FugIRAi8LAJ0eOVGO10FDjZcTMc7q24TXdfElDgCdHM0g l2ZanHEt/eX9yZTw/jVQxGc= =7Mz4 -----END PGP SIGNATURE-----

3 3

[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am not sure whether or not my procedure is right as server doesnt work at the moment, but I will keep working on it. Here is the Fedora installation [1] Fedora 6 linux install; [2] pre-dependencies package; Because I installed OpenIMSCore and Openser Presence module before, so some dependencies was already installed. In this case I just installed clearsilver from Add&Remove Package, and copy lxml folder(1.3.4 download from lxml site) to /usr/lib/python2.5/site-packages/. [3] install OpenXCAP, followed installation guide/Debain unstable installation By now, that's it. Coz no daemon can be run from /etc/init.d/openxcap. Also, I haven't modified anything in the config.ini and openxcap files Cheers, Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

1 0

[OpenSER-Users] OpenXCAP install and config problem?
by KevinKinnan 06 Sep '07

06 Sep '07

I am trying to install openXCAP on the Fedora 6. Here is my installation procedure: python ./setup.py install; copy config.init to /etc/openserxcap/ copy openxcap to /etc/init.d/ run /etc/init.d/openxcap start update openser config file modparam("presence_xml", "force_active", 0); modparam("presence_xml", "integrated_xcap_server", 0); I havent change anything in config.ini and openxcap it shows some errors "start-stop-daemon command not found". It seems that the openxcap.ini is writed for Debian based Linux as start-stop-daemon is for Debain. How can I run the start file on a red hat based linux? Do I need to do any other modification in some files? Thank You! Kevin _________________________________________________________________ MSN 中文网，最新时尚生活资讯，白领聚集门户。 http://cn.msn.com

2 1

[OpenSER-Users] Please help in Openser behind NAT design
by Halomoan Chow 06 Sep '07

06 Sep '07

Hi All I'm a new member and trying to use Openser as my VOIP "router" I plan to make following configuration: UA <---> Openser <---> Firewall/NAT <--INTERNET--> Firewall/NAT <---> UA a.a.a.a/24 x.x.x.x y.y.y.y b.b.b.b/24 a.a.a.a and b.b.b.b is internal IP not internet route-able x.x.x.x and y.y.y.y is Firewall/NAT external IP number I know there is mediaproxy and RTPProxy to solve the UA b.b.b.b that behind NAT but I don't know if my decision to put Openser behind NAT is correct or not. How to archive the above diagram with Openser ? If above Openser location is wrong, where should the Openser reside in the diagram? Install in Firewall/NAT or in INTERNET (dedicated server) Thank you for your help and sorry if my question is too newbie Halomoan

1 0

Re: [Serusers] Can't make a single call. Help me please.
by Mike Trest - Personal 06 Sep '07

06 Sep '07

As a long time VoIP and Internet engineer, we have all seen various port-blocking activities at different levels. These are almost always done to attain economic or political goals. (Remember PANAMA and SINGAPORE). Seldom is it ever for security reasons or for conservation of resources. It is the first question I ask whenever I start to assist a gateway operator who is testing via a cable operator. Therefore, it is understandable that we would be groping in the dark without that bit of bank ground information. Perhaps we should add a guideline note to our FAQ which recommends placing a sniffer/etherial between the originating sip device and the IP network as well as an ethereal/tcpdump on the SER end. ..mike.. At 01:34 AM 9/3/2007, Harmeet Singh wrote: >Dear Greg > >I tried ekiga, brujula, and couple other services on my internet >yesterday. None worked. Same problem - REGISTER is OK but can't make >calls. Then after searching google again, I found that my ISP (Rogers >Cable) is dropping/blocking the INVITE messages. That's why I was able >to register and not make any calls. Then I found on the internet about >a workaround, which is to not use 5060 port. So, I just tried that. To >my surprise, Any port other than 5060 just works! It beats me really. > >Thanks again for your help. I really appreciate it. > >Harmeet --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000771-1, 09/01/2007 Tested on: 9/6/2007 10:25:41 AM avast! - copyright (c) 1988-2007 ALWIL Software. http://www.avast.com

1 0

[OpenSER-Users] openSER rtpproxy video call - please help
by Volodymyr Fedyk 06 Sep '07

06 Sep '07

Hi, I've tried this openser configs from http://www.voip-info.org/wiki/view/OpenSER+v1.2.x+and+RTPProxy and generated by http://sipwise.com/wizard/ and I always face with the next issue: 2 UAs are behind NAT; each UA could call another; IM works; Audio works; but looks-like video streams are handled by openSER and rtpproxy incorrectly, and I'm unable to make a video call to any ua, - I'm using x-lit sp phone, and it shows "Could not start video" on one side, and "No matching video codec" on the other side... Has anybody faced with such issues later or has any ideas ? best regards, Dorin

3 2

[OpenSER-Users] Almost the End / End calls OpenSER + mediaproxy + freeradius
by Marc LEURENT 06 Sep '07

06 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello evrybody! I have one little problem... When a call badly ends (like hard disconnection or killall mysoftphone) mediaproxy has a timeout of 10sec, after the timeout, it sends a message to freeradius to stop accounting (so CDRTool billing too), but DON'T say to openser to stop the call, so the connection to my SIP/PSTN provider is still up (and its billing too) Is there any way to say to openser to send BYEs when mediaproxy detects the end of a badly ended call? Best Regards! PS: J-20 before VoN Europe 2007 - Rome Extract of the INSTALL file: """If the media session timeouts (there is no BYE that closes the session), MediaProxy will contact back the proxy dispatcher and pass the session media information to be logged. Using this architecture, only the proxy dispatcher needs to have direct access to the database. This addresses the issue of the MediaProxy servers being distributed and not having access to the database.""" -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3rpwqjpLE0HiOBYRAmbmAJ9UiDDi6UWYg4v+KLR020MwG/ks2ACggg8w aDQX87YSNxjEvy5vcSSeWto= =OCrH -----END PGP SIGNATURE-----

1 1

[OpenSER-Users] openser as proxy relay
by Asim Riaz 06 Sep '07

06 Sep '07

Hi, Could any body give me some idea to what need to be done in openser for relaying all SIP packets to another third party SIP registrar and Gateway. Scenario XYZ SIP Reg------------------openser----------------------UA Thanks in advance

3 2

[Serusers] serweb login problem
by Harmeet Singh 06 Sep '07

06 Sep '07

Hi I am facing this strange problem, if anyone can help me on this I will appreciate. Here is my configuration - Fedora 7, Apache 2.2.4, php 5.2.2, ser/serweb 0.9.6 When I go to localhost/serweb, enter login info, and submit, I get a download window pop-up asking me to save the php file. Shouldn't Apache be catching this and render the php to html? The login page itself is a php page and that renders ok. So, Apache is configured correctly for php rendering. What else am I missing here? Any help is appreciated. Thanks Harmeet

1 0

[OpenSER-Users] XCAP setting problem?
by Adrian Georgescu 05 Sep '07

05 Sep '07

You may obtain the XCAP server and the configurations from http:// openxcap.org The latest OpenSER svn version works with it and the sample configuration. Adrian >>>>> I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin

1 0

[OpenSER-Users] XCAP setting problem?
by KevinKinnan 05 Sep '07

05 Sep '07

The PUBLISH request works when using the following configurations(no XCAP): loadmodule "sl.so" loadmodule "mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "pidf_manipulation", 1) It doesn't if starting XCAP by updating these, the server can run properly, but cann't handle PUBLISH. loadmodule "xcap_client.so" modparam("presence_xml", "xcap_table", "xcap_xml") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_server", "localhost:5065") I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin _________________________________________________________________ Windows Live Spaces 中最年轻的成员！ http://miaomiaogarden2007.spaces.live.com/

1 0

[OpenSER-Users] mi_xmlrpc compilation issues
by Purna Chandar M 05 Sep '07

05 Sep '07

Hi, Im trying to compile mi_xmlrpc module in openser but could not succeed. The following errors are displayed mi_xmlrpc.c: In function 'xmlrpc_process': mi_xmlrpc.c:161: error: 'xmlrpc_server_abyss_rpc2_handler' undeclared (first use in this function) mi_xmlrpc.c:161: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:161: error: for each function it appears in.) mi_xmlrpc.c:167: error: 'xmlrpc_server_abyss_default_handler' undeclared (first use in this function) Im using Suse 10.1; The libraries xmlrpc-c-0.9.10-36, libxml-1.8.17-385, libxml2-devel-2.6.23-13, xmlrpc-c-devel-0.9.10-36 are properly installed. Few discussions on the mailing list concluded that the module mi_xmlrpc compiles well on Debian without any dependency issues. Is it true?? Can somebody tell me whether Im missing any dependencies?? Please let me know how to successfully compile this module. Thanks and regards, Purna Chandar M

3 3

Re: [OpenSER-Users] mi_xmlrpc compilation issues
by Kriti Wadhwa 05 Sep '07

05 Sep '07

2 1

[OpenSER-Users] NatPing stop working
by Marcello Lupo 05 Sep '07

05 Sep '07

Hi to all, i have openser 1.2.1 active from some months.. Suddenly on yesterday the natping stop to work. I have 140 contacts in user location almost all of them natted. There is some limit to user in location table at which the ping stops or a particular server load? There is a way to have something in the logs about it?? I didn't found anything in the logs. This is my config: modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) With a restart of openser all come back to work but i'm afraid if it will happen again? Thanks for help, Bye, Marcello

2 14

[OpenSER-Users] Opnser rejecting BYE packet
by Dan 05 Sep '07

05 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

4 19

Re: [OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?
by David Loh 05 Sep '07

05 Sep '07

Hi, Arrggghh .. that's one of my attempts to eliminate the broken "BYE" problem... that's ngrep was captured when I set "modparam("rr", "enable_double_rr", "0");", I've paste another ngrep to http://pastebin.ca/674450, this time the double RR header is enabled. And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt the post is "openser"). Even though double RR header is enabled, but for BYE it's still doesn't process properly :( For the .cfg file line #130 onward, I did tried t_relay, forward and force_send_socket, but none of this will do the trick (force_send_socket was complaining TLS error due to missing certificate (?) ) Would appreciate if anyone could enlighten me why is this happen ? Thanks, David Loh Klaus Darilion wrote: > But the INVITE you posted at http://pastebin.ca/673392 also has only > one Record-Route header. > > regards > klaus > > David Loh schrieb: >> Hi, >> >> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK >> ...but when asterisk disconnected the call and send BYE back to OpenSER, >> the TLS RR header wasn't present, the only 2 RR header was >> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >> I'm puzzled ... is there any command to 'fix' this? >> >> >> Regards, >> David Loh >> >> Klaus Darilion wrote: >>> The openser proxy should add 2 record-route header (TLS and UDP = >>> double record route). This is why it does not work. >>> >>> regards >>> klaus >>> >>> David Loh schrieb: >>>> Hi All, >>>> >>>> Greeting. >>>> >>>> I've been struggle with OpenSER TLS implementation for more than a >>>> week, since I've ported from UDP to TLS, everything work fine >>>> except the "BYE" request from Asterisk (loose route), my >>>> implementation was something like below: >>>> >>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>> >>>> My OpenSER.cfg already configured to listen on two port which is :- >>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or >>>> even voicemail) having no problem, >>>> but when the callee disconnect the call, caller will never get hang >>>> up :( >>>> >>>> I've attached my ethereal trace/ngrep to pastebin, >>>> http://pastebin.ca/673392 >>>> >>>> Wondering if anyone can help me with the broken "BYE" that returned >>>> from Asterisk ? >>>> Line #131, supposedly this line should have contain 2 Via header, >>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>> but somehow the TLS via header was gone !! (compare to previous ACK >>>> (Line #117) /INVITE (Line #51). >>>> Due to the missing TLS via header, OpenSER log file was complaining >>>> "protocol/port mis-match". >>>> >>>> The last BYE request (Line #256) is actually firing from Client, >>>> which contain the "TLS" via. >>>> >>>> >>>> I've even tried "force_send_socket" to port 5061 (instead of 5060) >>>> from loose route, but it complaining TLS certificate error, >>>> since Asterisk doesn't support TLS natively, I've no clue why is >>>> the ACK/INVITE/CANCEL work but not BYE. >>>> if (loose_route) { >>>> .... >>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>> } >>>> >>>> >>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk >>>> setup, >>>> I'm really appreciate if you can share your experience with me, or >>>> pin point what's the mistakes I made here. >>>> >>>> Thanks in advance. >>>> >>>> Regards, >>>> David Loh >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > >

2 5

[OpenSER-Users] OpenSER with PostgreSQL
by Cliff Flood 04 Sep '07

04 Sep '07

Hi, I have an Ubuntu 6.06 install with PostgreSQL 8.1 and OpenSER 1.2.2 which I am testing at the moment. I have tested OpenSER with MySQL and it worked fine but when when I want to switch to PostgreSQL I'm having some difficulty. I have used the postgressqldb.sh script to initialise the database and it has completed without error. When I start OpenSER I get the following lines in my PostgreSQL logs: LOG: connection received: host=127.0.0.1 port=37610 LOG: connection authorized: user=openser database=openser LOG: SSL SYSCALL error: EOF detected LOG: could not receive data from client: Connection reset by peer LOG: unexpected EOF on client connection I have looked at OpenSER's logs with debugging turned on but I don't see anything that looks like an error but I would be happy to provide them. `openserctl moni` returns: ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. My openser.cfg looks like this: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 listen=192.168.20.151 dns=no rev_dns=no port=5060 mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; 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=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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("host.doman.tld", "subscriber")) { # www_challenge("host.domain.tld", "0"); # exit; # }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; } Any and all help appreciated. -- Cliff Flood Systems Administrator Jazinga Inc.

2 2

[OpenSER-Users] Strange Provider SIP - DSP fields in trying
by Marc LEURENT 04 Sep '07

04 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...) When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced! OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP! Do you know how I can delete or modify TRYING/RINGING SDP? Thanks Below an example of a RINGING with SDP # U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712(a)192.168.95.47. Contact: <sip:212.129.6.65:5060>. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903. Record-Route: <sip:88.191.45.91;lr;ftag=542924903>. Server: Cirpack/v4.39a (gw_sip). To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TdoqjpLE0HiOBYRAjnkAKCgzB4p89oTKhxuvZcLlklTme9w3ACeP+TW FI+LkwgbaJ1XJTIm3FUSMTg= =Cp53 -----END PGP SIGNATURE-----

2 4

[OpenSER-Users] MediaProxy 1.9.0
by Mike O'Connor 04 Sep '07

04 Sep '07

Hi Guys I upgraded my mediaproxy to 1.9.0 and found out sometime later that the data usage update at call end seems to have stopped. I'm using direct mysql (not radius) for the end of call updates. No error message in the logs and a manual test of mysql show nothing wrong. Any idea. Mike

1 0

RE: [OpenSER-Users] Opnser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Hi Ravi, Did what you suggested, but still same error. Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 3:51 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Dan In yur openser.cfg In Below this this Section route { if(loose_route()) { # mention here BYE section also with fix_nated_contact(); } if(method=="CANCEL" ) { } else if (method=="INVITE") { } else if (method=="REGISTER") { } else if ( method=="BYE" ) # here u missing in this section route(1); exit; } On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: Where exactly? Regards Dan _____ From: raviprakash sunkara [mailto:sunkara.raviprakash.feb14@gmail.com] Sent: Tuesday, September 04, 2007 2:27 PM To: Dan Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hi Dan Problem in ur openser,cfg file On 9/4/07, Dan <fiedler.dan(a)gmail.com> wrote: My Config File: listen=72.65.56.136 #alias=0.0.0.0 alias=dstpl.com check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=6 fifo="/tmp/openser_fifo" fifo_db_url="postgres://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/postgres.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/uac.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/textops.so" # Uncomment this if you want diest authentication mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_radius.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/avp_radius.so" loadmodule "/usr/local/lib/openser/modules/group_radius.so" # loadmodule "/usr/local/lib/openser/modules/lcr.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" #loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "postgres://openser:openserrw@localhost/openser") # ----------------- setting module-specific parameters --------------- # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) modparam("maxfwd", "max_limit", 32) modparam("auth_radius","radius_config","/etc/radiusclient-ng/radiusclient.co nf") modparam("auth_radius", "service_type", 15) modparam("avp_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- Accounting param -- modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "service_type", 16) modparam("acc", "radius_flag", 1) modparam("acc", "log_level", 1) # modparam("acc", "radius_missed_flag", 1) modparam("acc", "report_ack", 0) # 1 reporta dos starts en acc modparam("acc", "radius_extra", "Sip-Source-IP-Address=$si") # -- 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) # modparam("avpops", "avp_aliases", "c_uri=i:888; s_ip=i:999" ) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if((method=="INVITE") || (method=="ACK") || (method=="BYE") || (method=="CANCEL")) { setflag(1); } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len > 2048 ) { sl_send_reply("513", "Message too big"); 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" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); exit; }; if (!uri==myself) { route(5); route(1); exit; }; if (uri==myself) { if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="BYE") { route(1); exit; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); 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(); force_rport(); }; sl_send_reply("100", "Trying"); if (!radius_www_authorize(" <http://72.65.56.136> 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!save("location")) { 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(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[3] { # ------------------------------------------------------------------------ # INVITE Handler # ------------------------------------------------------------------------ if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(5); route(1); exit; }; if (!allow_trusted() && uri=~"^sip:[0-9]*@.*") { if (!radius_www_authorize(" 72.65.56.136")) { www_challenge("72.65.56.136", "0"); exit; }; }; if (uri=~"^sip:[0-9]*@.*"){ route(11); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; if (method=="CANCEL") { route(1); exit; }; route(5); route(1); } route[11]{ # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- strip(0); prefix(""); rewritehostport(" 206.234.108.104:5060 <http://206.234.108.104:5060> "); route(5); route(1); } -----Original Message----- From: users-bounces(a)openser.org [mailto: <mailto:users-bounces@openser.org> users-bounces(a)openser.org] On Behalf Of Dan-Cristian Bogos Sent: Tuesday, September 04, 2007 1:30 PM To: users openser.org Subject: Re: [OpenSER-Users] Opnser rejecting BYE packet Hello Inaki, yes, it is almost sure that it does (except receiving this answer from remote party). This is the reason all the people place loose-route block before location query and this is why I have assumed that it does not match the loose-routing and ends up to location query automatically. DanB On 9/4/07, Marc LEURENT <lftsy(a)free.fr> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Like Dan said, > send us your configuration, you may have forbidden something! > > Dan a écrit : > > *_Scenario 1:_* > > > > > > > > > > > > U 212.34.169.220:5060 -> 78.45.146.34:5060 > > > > BYE sip:test@78.45.146.34 SIP/2.0. > > > > Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Content-Length: 0. > > > > > > > > # > > > > U 78.45.146.34:5060 -> 212.34.169.220:5060 > > > > SIP/2.0 404 User Not Found. > > > > CSeq: 2 BYE. > > > > Via: SIP/2.0/UDP 212.34.169.220:5060. > > > > From: < <mailto:sip:1217773456@78.45.146.34> sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. > > > > Call-ID: 8246bf729f62a754. > > > > To: 58.68.96.11< sip:test@78.45.146.34 <mailto:sip:test@78.45.146.34> >;tag=aa313a3c. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 > > req_src_ip=78.45.146.34 req_src_port=5060 > > in_uri=sip:78.45.146.34;ftag=aa313a3c;lr > > out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". > > > > > > > > > > > > *_Scenario 2:_* > > > > > > > > U 59.176.78.110:50496 -> 210.68.65.78:5060 > > > > BYE sip:210.68.65.78 <http://210.68.65.78> SIP/2.0. > > > > Route: <sip:@209.67.171.10;transport=udp>. > > > > Via: SIP/2.0/UDP 59.176.78.110:50496 . > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > <sip:12818495027@210.68.65.78 ;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > User-Agent: Cisco ATA v2.15 ata18x (020927a). > > > > Content-Length: 0. > > > > . > > > > > > > > # > > > > U 210.68.65.78:5060 -> 59.176.78.110:50496 > > > > SIP/2.0 478 Unresolvable destination (478/TM). > > > > Via: SIP/2.0/UDP 59.176.78.110:50496. > > > > From: sip:user1@210.68.65.78;tag=4154163258. > > > > To: > > < <mailto:sip:12818495027@210.68.65.78> sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. > > > > Call-ID: 4202237689(a)192.168.1.156. > > > > CSeq: 3 BYE. > > > > Server: OpenSer (1.0.1 (i386/linux)). > > > > Content-Length: 0. > > > > Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 > > req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. > > > > > > > > > > > > In both cases BYE is not getting accounted for. Can anyone help me on > > this ?? > > > > > > > > > > > > Regards > > > > Dan > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org <http://enigmail.mozdev.org> > > iD8DBQFG3Q9/qjpLE0HiOBYRArgmAKCec0eFK2QQyBmV5eGoCWRmUOMM8ACgmFgp > tOKvu9D6afXK2f8if8FABnM= > =66HB > -----END PGP SIGNATURE----- > > _______________________________________________ > 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 _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users -- Thanks &Regards Ravi Prakash Sunkara -- Thanks &Regards Ravi Prakash Sunkara

1 0

[Serusers] SER+PATTON1+PATTON2+PATTON3
by pokerino76 04 Sep '07

04 Sep '07

I do not have the possibility to use Patton 4960 with 1xPRI. I wished an alternative method for Patton1 + patton2 + patton3: 8+8+8=32. thanks This is the code for 2 Patton Hi! On the pattons you can created Error Messages when not circuit channel is available And on Ser you have to register to Failure Route to the route of the first patton and the failure route point to the second Patton. PATTON1: profile sip default map causes to-sip no-circuit-channel-available to 486 map causes to-sip resources-unavailable to 486 service hunt-group HUNT-BRI cyclic timeout 1 drop-causes normal-unspecified drop-causes no-circuit-channel-available drop-causes network-out-of-order drop-causes temporary-failure drop-causes switching-equipment-congestion drop-causes access-info-discarded drop-causes circuit-channel-not-available drop-causes resources-unavailable route call 1 dest-interface ISDN_01 route call 2 dest-interface ISDN_02 route call 3 dest-interface ISDN_03 SER: Ser.cfg: if (uri=~ ". *sip: 012345 ") # Patton 1 {t_on_failure (1); t_relay_to_udp (11.11.11.10, 5060); break; } failure_route [1] { if (method== " INVITE ") { if (t_check_status (600|486)) #### ErrorCode from Patton 1 { #####call from Patton 1 -- 486 (Busy) ret -- call switch to Patton 2 ###### if (uri=~ ". *sip: 012345 ") # Costumer xy { append_branch (); t_relay_to_udp (11.11.11.11, 5060); ##Unterroute ist hier nicht notwendig, break; ##da sich die Verrechnung hier nicht aendert } } } Which code to use for 3 patton? ------------------------------------------------------ Leggi GRATIS le tue mail con il telefonino i-mode di Wind http://i-mode.wind.it/

1 0

[OpenSER-Users] Openser rejecting BYE packet
by Dan 04 Sep '07

04 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

2 2

[OpenSER-Users] call duration limit
by Nick 'TARANTUL' Novikov 04 Sep '07

04 Sep '07

Hi, ppl! I have installed OpenSER and radius server. OpenSER authorized all call through radius. How I can limit call duration by "Session-Timeout" attribute from radius ? -- tarantul

7 10

[Serusers] Route SIP Calls To Asterisk
by Kevin Kibuku 03 Sep '07

03 Sep '07

Hi Team, Please someone guide me in how I could forward for instance a call to 1111111 made on the SER to an * server. I have tried quite a lot of things but nothing seems to work. Thanks -- Kevin Kibuku, Intersat Africa Limited Internet Via Satellite Cell: +254720789158

2 2

[OpenSER-Users] t_relay after failure_route -- problem
by Zahid Mehmood 03 Sep '07

03 Sep '07

Hi All, I am trying to implement serial forking and am using the example on voice-systems website as a guide. The problem I am experiencing is as follows. Failure route rewrite ruri with the next uri to try and calls route[17] route 17 performs a lookup("location"), finds the usrloc information and rewrites ruri, however, when t_relay is called it uses the original ruri (one before the lookup) to do t_relay. Another issue I'm experiencing is that when processing call after failure route, ruri_user_avp. Once i rewrite the uri in failure route, I have to use the following to manually write a value for ruri_user_avp. Can someone explain why this is happening? Thanks in advance for your help!! -- Zahid ] Some relevant config and debug info: modparam("lcr", "gw_uri_avp", "1400") modparam("lcr", "ruri_user_avp", "1402") route[18] { # used to process pstn/ off system calls ...... if (!is_avp_set("$avp(i:1402)")) { xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp **\n"); $avp(i:1402) = $rU; xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp = $avp (i:1402) **\n"); } ........ } #end route 18 route[11] { # load all numbers asscociated with the pilot number in $avp (serial_fork) array xlog("L_INFO","$ci: route[11] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(serial_fork) = "sip:*6" + $rU + "@sipdev.oursiphost.com"; if (avp_db_query("select username from fork where pilot = $rU order by pri" , "$avp(serial_fork)")) { xlog("L_INFO", "route[11] $ci: method $rm r-uri <$ru> \n"); xlog("L_INFO", "route[11] serial_fork = $avp (serial_fork) remote-user = $rU \n"); t_on_failure("5"); route(17); exit; } else { xlog("L_ERR", "route[11] failed to load serial_fork info.\n"); } avp_print(); # debugging only setflag(8); } # end of route[11] route[17] { xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si \n"); $avp(s:inv_timeout) = "10"; if (lookup("location")) { # currently online registered UA. xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- inside location lookup \n"); if (!t_relay()) { sl_reply_error(); }; return; } else { # not online if (does_uri_exist()) { # but exists xlog("L_INFO", "$ci: Subscriber is offline [$ru]\n"); sl_send_reply("480", "Temporarily Unavailable."); return; } xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si -- outside location lookup \n"); }; if (uri=~"^sip:(([0-9()+-])|(%20))+@") { subst_user('/[()+-]//g'); # strip out the punctuation. subst_user('/%20//g'); # strip out the white space. route(18); # all numerics just go to route(18) -- off system calls return; }; } # end route[17] failure_route[5] { xlog("L_INFO","$ci: failure_route[5] - M=$rm RURI=$ru F=$fu T=$tu IP=$si\n"); avp_print(); if (avp_pushto("$ruri", "$avp(serial_fork)")) { append_branch(); xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); avp_delete("$avp(serial_fork)"); t_on_failure("5"); route(17); } else { xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru> from <$fu> to <$tu>\n"); } } Debug info Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] - M=INVITE RURI=sip: 10512@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039c28, flags=0x0003 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<inv_timeout> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<10 / 2> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bf0, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10511@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039bb8, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10210@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b80, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:10510@sipdev.oursiphost.com / 29> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b48, flags=0x0002 Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<sip:*610512@sipdev.oursiphost.com / 31> Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp: p=0xb6039b18, flags=0x0083 Aug 31 10:54:56 mousse openser[6815]: INFO: name=<is_pilot> Aug 31 10:54:56 mousse openser[6815]: INFO: val_str=<y / 1> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:pushto_avps: 1 avps were processed Aug 31 10:54:56 mousse openser[6815]: ******* setting for branch 0 flags 0 Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: failure_route[5] INVITE r-uri <sip: 10511(a)sipdev.oursiphost.com> from <sip:10510@sipdev.oursiphost.com> to <sip:10512@sipdev.oursiphost.com;user=phone> Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:delete_avps: 1 avps were removed Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511(a)sipdev.oursiphost.com F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 Aug 31 10:54:56 mousse openser[6815]: rewrite_uri: Rewriting Request- URI with 'sip:10511@10.1.12.172:8891' Aug 31 10:54:56 mousse openser[6815]: a410a3c1- e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip: 10511@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip: 10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 -- inside location lookup Aug 31 10:54:56 mousse openser[6815]: DEBUG: mk_proxy: doing DNS lookup... Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: no port, no proto -> do NAPTR lookup! Aug 31 10:54:56 mousse openser[6815]: DEBUG:filter_and_sort_naptr: found valid SIP+D2U -> _sip._udp.sipdev.oursiphost.com Aug 31 10:54:56 mousse openser[6815]: DEBUG:do_srv_lookup: SRV (_sip._udp.sipdev.oursiphost.com) = mousse.cc.oursiphost.com:5060 Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: found! Aug 31 10:54:56 mousse openser[6815]: check_via_address(10.1.12.174, 10.1.12.174, 0) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039eb4 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039ed0 (91) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:relay_reply: branch=0, save=1, relay=-1 >, CSEQ_N=<CSeq: 1> 2(a)sipdev.oursiphost.com;user=phone>ocal: using FROM=<From: "Zahid 10510" <sip: 10510(a)sipdev.oursiphost.com>;tag=BF787F6-F6DD44ED Aug 31 10:54:56 mousse openser[6815]: DEBUG: cancel_branch: sending cancel... Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 500000 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]: 0xb6039e20 (61500000) Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative timeout is 30 Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]: 0xb6039e3c (91)

1 2

[OpenSER-Users] Bernard Le est absent(e).
by bernard.le＠orange-ftgroup.com 03 Sep '07

03 Sep '07

Je serai absent(e) du 03/09/2007 au 17/09/2007. Je répondrai à votre message dès mon retour. Contacter le service Ingénierie : Patrice Galleau au +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com ou Tony Capochichi au +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com I will reply to your message when I'll be back. Please contact the Engineer department: Patrice Galleau at +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com or Tony Capochichi at +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com ********************************* This message and any attachments (the "message") are confidential and intended solely for the addressees. Any unauthorised use or dissemination is prohibited. Messages are susceptible to alteration. France Telecom Group shall not be liable for the message if altered, changed or falsified. If you are not the intended addressee of this message, please cancel it immediately and inform the sender. ********************************

1 0

[OpenSER-Users] Does Presence Module support 'policy document'?
by KevinKinnan 03 Sep '07

03 Sep '07

Does Presence Module support 'policy document'? That means the user can do some settings to forbid specified watcher or something like this. Thank you! Kevin _________________________________________________________________ Windows Live Custom Domain，您的免费电子邮局。 https://domains.live.com/default.aspx

2 1

[OpenSER-Users] OpenXCAP launch
by Adrian Georgescu 03 Sep '07

03 Sep '07

Hello, I would like to announce the launch of OpenXCAP.org OpenXCAP is an open source, easy extensible, fully featured XCAP server with TLS security and support for multiple realms. OpenXCAP server can be used in combination with a SIP server that supports PUBLISH/SUBSCRIBE/NOTIFY methods to provide a complete SIP SIMPLE server solution The server is written in the Python programming language and is based on the following RFCs: - The Extensible Markup Language Configuration Access Protocol RFC 4825 - Formats for Representing Resource Lists RFC 4826 - Usage for Manipulating Presence Document Contents RFC 4827 - Presence Authorization Rules draft-ietf-simple-presence-rules-10 The server supports TLS security, multiple back-end storage systems and works out of the box with OpenSER Proxy/Registrar/Presence server. The software is provided under the BSD license. I wish to thank here to my IETF friends who helped me with advice, testing and guidance during the development phase of this project. I would also like to thank to Mircea Amarascu, the author of the software, without his passion and dedication this project would not have been possible. The project is hosted at: http://openxcap.org The project is supported via the wiki collaboration system setup by AG Projects. To open ticket please Register first. We are looking forward your feedback from real-life field deployments. Kind regards, Adrian Georgescu AG Projects

2 1

[OpenSER-Users] Media control with openser
by srinivas Antarvedi 03 Sep '07

03 Sep '07

Hello all, Actually i have openser box which i used to make calls among my offices which spread different geographical places. i am using in the configuration file just to make changes to my rtp packets so without RTPPROXY and MEDIAPROXY modules i was able to make calls even my end UAC are behind NAT. but if i want to control my media between different UAs like a prepaid system as we used to see in case of mobile prepaid system which stops you from continuing your call when there was no balance left on ur mobile . can anybody give me a hint how can i do that ? if so what architecture it needs like any mediaproxy? will session border controller help me out? just give me a hint so that i can go ahead thanks once again -- Srinivas Antarvedi

3 2

[OpenSER-Users] Wanted: VoIP Engineer for Warsaw!
by laurent schweizer 03 Sep '07

03 Sep '07

Peoplefone AG jest firmą oferującą usługi telekomunikacyjne oparte na technologii Voice over IP (VoIP) w wyjątkowo niskich stawkach. Peoplefone jest certyfikowanym partnerem firm Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>i AVM/FRITZ!Box <http://www.fritz-shop.ch/>. W związku z naszym dynamicznym rozwojem, do nowo tworzonej spółki w Polsce poszukujemy osoby na stanowisko: SPECJALISTA D/S VOIP Miejsce pracy: Warszawa *Wymagania:* - wykształcenie wyższe techniczne (lub w trakcie studiów), najlepiej informatyczne - praktyczna umiejętność programowania w środowisku w PHP i językach C i C++ - znajomość Linux - znajomość mySQL i PostgreSQL - znajomość zagadnień związanych z sieciami IP, UDP, TCP - znajomość narzędzi do analizy sieci np.. Ethereal - wiedza z zakresu VoIP, serwerów i konfiguracji oprzyrządowania - umiejętność rozwiązywania problemów informatycznych - znajomość języka angielskiego w mowie i piśmie w stopniu zaawansowanym - umiejętność szybkiego uczenia się - chęć podnoszenia swoich kwalifikacji zawodowych - profesjonalne i rzetelne podejście do obowiązków służbowych, umiejętność pracy w zespole, samodzielność, komunikatywność, systematyczność, terminowość - doświadczenie zawodowe mile widziane *Dodatkowe wymagania (mile widziane):* - umiejętność programowania w PERL i JAVA - znajomość Asterisk / SER / OPENSER - umiejętność konfigurowania bramek i ruterów Cisco, Patton, lub innych - znajomość protokołów SIP lub STUN - znajomość problematyki NAT I funkcjonowania outband Proxy - znajomość narzędzi monitorujących jak Cactus, Nagios, MRTG - znajomość narzędzi jak DRBD, Hearthbeat - dyspozycyjność Wszystkich zainteresowanych prosimy o przesłanie życiorysu oraz listu motywacyjnego (zawierającego zgodę na przetwarzanie danych osobowych) *do 10 września br. ** *na adres: *ewa.ciesla(a)peoplefone.com* W temacie wiadomości wysyłanej pocztą elektroniczną prosimy umieścić nazwę stanowiska. Skontaktujemy się tylko z wybranymi kandydatami. Prosimy o dopisanie następującej klauzuli "Wyrażam zgodę na przetwarzanie moich danych osobowych zawartych w mojej ofercie pracy dla potrzeb niezbędnych do realizacji procesu rekrutacji (zgodnie z ustawą z dn. 29.08.97 roku o Ochronie Danych Osobowych Dz. Ust Nr 133 poz. 883)".

1 0

Re: [Serusers] Can't make a single call. Help me please.
by Harmeet Singh 03 Sep '07

03 Sep '07

It's strange, but I do see the SIP INVITE messages being sent in the ethereal log. Why they don't reach to the server? SER is connected directly to the internet, without any NAT. On the other hand, REGISTER messages do get to SER and the client recieves the 200 OK and all intermediate replies as well. I have attached the clientside ethereal log for you to see. I am lost. Harmeet

3 3

[Serusers] exec_dset
by Jesús Terán 02 Sep '07

02 Sep '07

Hello, I am using the exec_dset for access to my database mysql and change the uri in the calls, but I think this use more CPU processing, Are there other module to do the same? thanks Jesus Teran B.

4 7

Re: [OpenSER-Users] OpenSER to PSTN
by Ricardo.ferreira 02 Sep '07

02 Sep '07

1 0

[OpenSER-Users] OpenSER to PSTN
by Arya 01 Sep '07

01 Sep '07

Hi Everyone Is there any way to send all the calls that come to OpenSER to PSTN lines (PRI, BRI) ? -- Thank You

2 1

Re: [OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi Dan, Thanks for all your help and support, I really appreciate it. I was able to get it to work. Although I had to replace all VSA'a with the SIP-AVP attributes. For some reason the SIP-AVP's were not loaded due to the presence of other VSA's. Maybe it was because I was using a = sign in the value prefixes instead of : or #. Is this a known issue. Regards, Danish ps: Someone asked earlier about the radius server I was using. The radius server along with the billing solution is from advancedvoip.com, so far our experience with them has been quite good. > Hi Danish, > > sorry for replying to you so late, but I was out from the office today. > > Here you go: > > First of all, my bad, the returned value should be in the form: > return-code#0 if you want your value to be treated as an integer or > return-code:0 if 0 the AVP should be considered as string - has been a > while since I worked last time with those. By using a higher layer of > debug you should be able to see your avps converted inside when > received from radius. > The rest of your configuration looks good. Note that you can return > more AVP values with the same reply, so you can include the credit > amount as well: > SIP-AVP="return-code#0" > SIP-AVP="h323-credit-amount#4206" > > Hope you will rock now. > > Cheers, > DanB > > On 8/30/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> > wrote: >> Hi Dan, >> To test the functionality of SIP-AVP, I am trying to get theses avp's in >> the registration block, shown below. Basically on sucessful registration >> the radius server returns the following attributes in acces-accpet >> >> Attributes: >> SIP-AVP = "return-code=0" >> cisco-h323-credit-amount = "h323-credit-amount=4206" >> >> so after registration I am trying to get the return code value. The >> registration block is shown below >> >> route[2] { >> # ----------------------------------------------------------------- >> # REGISTER Message Handler >> # ---------------------------------------------------------------- >> xlog("L_NOTICE","Danish: Entering route[2]\n"); >> xlog("L_NOTICE","Danish: Register message IPsrcaddr [$si], >> RecvdIPaddr[$Ri]\n"); >> if (!search("^Contact:[ ]*\*") && nat_uac_test("2")) { >> xlog("L_NOTICE","Danish: In route[2] Found Nated >> contact\n"); >> setflag(6); >> setbflag(6); >> # if you want OPTIONS natpings uncomment next >> setbflag(7); >> fix_nated_register(); >> force_rport(); >> }; >> >> sl_send_reply("100", "Trying"); >> if (!radius_www_authorize("192.168.0.99")) { >> www_challenge("192.168.0.99", "1"); >> }; >> consume_credentials(); >> >> #avp_print(); >> xlog("L_NOTICE","Danish: Before is set check\n"); >> if(is_avp_set("$avp(s:return-code)")){ >> xlog("L_NOTICE","Danish: return code is set \n"); >> }; >> xlog("L_NOTICE","Danish: After is set check\n"); >> if(avp_check("$avp(s:return-code)", "eq/i:0/g")) >> { >> xlog("L_NOTICE","Danish: return code 0 received \n"); >> }; >> >> if (!save("location")) { >> sl_reply_error(); >> }; >> >> } >> >> In my logs I see >> >> 058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair 225 >> 4(11058) Danish: Before is set check >> 4(11058) Danish: After is set check >> 4(11058) DEBUG:avpops:ops_check_avp: no src avp found >> >> so both conditional avp checks fail and the last debug line is a bit >> worrying >> I traced the first debug line above to sterman.c->generate_avps() >> so just under this debug line I added this piece of code >> >> vp=rc_avpair_get(vp,attrs[A_SIP_AVP].v,0); >> if (vp != NULL) >> { >> DBG("DEBUG:auth_radius:generate_avps: strvalue %s\n", >> vp->strvalue); >> } >> else >> DBG("DEBUG:auth_radius:generate_avps: vp is null\n"); >> >> and this is what I got >> >> 4(11058) DEBUG:auth_radius:generate_avps: getting SIP AVPs from avpair >> 225 >> 4(11058) DEBUG:auth_radius:generate_avps: vp is null >> >> hence openser is not loading the SIP-avp in $avp(s:return-code). I think >> either I am missing something in the openser script or the way I am >> returning the attribute. >> >> all relevant definitions in the cfg are given below: >> >> loadmodule "/usr/local/lib/openser/modules/auth_radius.so" >> loadmodule "/usr/local/lib/openser/modules/uri_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avp_radius.so" >> loadmodule "/usr/local/lib/openser/modules/avpops.so" >> >> modparam("auth_radius|uri_radius|avp_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") >> >> modparam("uri_db|usrloc", "db_url", >> "mysql://openser:openserrw@localhost/openser") >> >> modparam("nathelper", "natping_interval", 20) >> modparam("nathelper", "ping_nated_only", 1) >> modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") >> modparam("nathelper", "received_avp", "$avp(i:42)") >> modparam("nathelper", "sipping_bflag", 7) >> modparam("nathelper", "sipping_from", "sip:pinger@abc.org") >> >> modparam("registrar", "received_avp", "$avp(i:42)") >> modparam("usrloc", "db_mode", 2) >> modparam("usrloc", "nat_bflag", 6) >> >> Hope this explains everything and you would be in a better position to >> help. >> >> Regards, >> Danish >> ps: isnt there any sample script I can refer to for sip-avp radius >> configurations. >> >> >> > Hi Danish, >> > >> > can u post the full block where you are doing these checks? >> > >> > DanB >> > >> > On 8/29/07, danish.samad(a)vocalseeds.com <danish.samad(a)vocalseeds.com> >> > wrote: >> >> Hi Dan, >> >> Thanks for the explanation, things are more clear now. Now I am >> >> returning >> >> a SIP-AVP from radius (for authentication), this is what I see in the >> >> logs >> >> >> >> Attributes: >> >> SIP-AVP = "return-code=0" >> >> ....... >> >> >> >> In my script I do >> >> avp_print(); >> >> if(is_avp_set("$avp(s:return-code)")){ >> >> xlog("L_NOTICE","return code is set \n"); >> >> }; >> >> if(avp_check("$avp(s:return-code)", "eq/0/g")) >> >> { >> >> xlog("L_NOTICE","return code 0 received \n"); >> >> }; >> >> >> >> none of the checks pass and nothing is printed from avp_print. In >> >> openser >> >> logs I see this line >> >> >> >> 1(32550) DEBUG:auth_radius:generate_avps: getting SIP AVPs from >> avpair >> >> 225 >> >> >> >> am I missing some pre-requisites or what am I doing wrong. >> >> >> >> Regards, >> >> Danish >> >> >> >> >> >> > Hi Danish, >> >> > >> >> > for the moment you cannot process attributes other than SIP-AVP >> inside >> >> > openser script. This one is automatically converted into avp. >> >> > EG. A SIP-AVP attribute in the form of: "reason='No Credit'" should >> >> > automatically create an AVP named reason with value "No Credit". >> >> > >> >> > Hope that helps, >> >> > DanB >> >> > >> >> > >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> <danish.samad(a)vocalseeds.com> >> >> > wrote: >> >> >> Hi Dan, >> >> >> >> >> >> Thanks for your reply. >> >> >> >> >> >> Actually I am currently returning an integer status value in >> >> >> cisco-h323-return-code (a vendor specific attribute). So basically >> >> how >> >> >> can >> >> >> I extract the value returned in this attribute, compare it and >> then >> >> send >> >> >> and sl reply. >> >> >> >> >> >> Probably there is a way of loading this attr value in an avp and >> >> >> comparing >> >> >> it? >> >> >> >> >> >> Regards, >> >> >> Danish >> >> >> >> >> >> > Hello Danish, >> >> >> > >> >> >> > a solution would be returning a SIP-AVP, containing the reason >> for >> >> >> > failure, and then forward this to the client using sl. >> >> >> > Another one would be completely rewriting R-URI again from your >> >> >> > SIP-AVP and then send the request to an announcement server (eg: >> >> >> > ann:paymentrequired@your.server - which should play Payment >> >> Required >> >> >> > annoucement). >> >> >> > >> >> >> > Cheers, >> >> >> > DanB >> >> >> > >> >> >> > On 8/29/07, danish.samad(a)vocalseeds.com >> >> <danish.samad(a)vocalseeds.com> >> >> >> > wrote: >> >> >> >> Hi, >> >> >> >> >> >> >> >> I have configured openser-1.2 with a (commercial) third party >> >> radius >> >> >> >> server and it is working great. Now I have a requirement where >> I >> >> need >> >> >> to >> >> >> >> check a radius return attribute in case an access reject is >> >> received >> >> >> in >> >> >> >> response to an authorization request. Is there some way I can >> do >> >> >> this >> >> >> >> in >> >> >> >> the openser script. >> >> >> >> >> >> >> >> Basically I want to send a 402 payment required message to a >> sip >> >> >> client >> >> >> >> in >> >> >> >> case the call is rejected due low balance. In such cases radius >> >> >> returns >> >> >> >> a >> >> >> >> status response code with access reject. >> >> >> >> >> >> >> >> >> >> >> >> Thanks, >> >> >> >> Danish >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Users mailing list >> >> >> >> Users(a)openser.org >> >> >> >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> >> >> > >> >> >> >> >> >> >> >> > >> >> >> >> >> > >> >> >

1 0

[OpenSER-Users] checking radius return attributes
by danish.samad＠vocalseeds.com 01 Sep '07

01 Sep '07

Hi, I have configured openser-1.2 with a (commercial) third party radius server and it is working great. Now I have a requirement where I need to check a radius return attribute in case an access reject is received in response to an authorization request. Is there some way I can do this in the openser script. Basically I want to send a 402 payment required message to a sip client in case the call is rejected due low balance. In such cases radius returns a status response code with access reject. Thanks, Danish

4 10

Re: [Serusers] INFO Method
by Greger Viken Teigre 01 Sep '07

01 Sep '07

As I wrote: you don't, you need another component to do that. SER has a management interface that can be used to inject a message, but you still need an external component to do that. g-) ------- Original message ------- From: kapilg <kapilg(a)amdale.com> Sent: 31.8.'07, 13:34 > Oh Sorry! Actually I am using SER. > > > Well, if one has to send INFO using ser then how one can send? > > > > > Regards, > > > Kapil > > > > --------------- > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Friday, August 31, 2007 4:19 PM > To: kapilg > Cc: serusers(a)iptel.org > Subject: Re: [Serusers] INFO Method > > > > > Wrong list, this is the ser, not openser list. > A proxy just proxies, it doesn't create messages. But maybe your media server? > g-) > > kapilg wrote: > > > Dear Users, > > > > > Could any one tell or give me one sample of sending INFO Method with MSCML as Message Body part? > > > I am working on one project in which XLite makes a call to a Media Server using openser proxy and after some seconds delay proxy forms one packet of INFO Method filling MSCML script in this and send to Media Server. This will be an ivr call. > > > > > Any help will be appreciated. > > > > > Thanks and Regards, > > > Kapil Goel > > > > > > --------------- > > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] Can't make a single call. Help me please.
by Harmeet Singh 31 Aug '07

31 Aug '07

Hi allAfter spending whole week reading and googling I am writing here. Hope someone from here can help me.Here is my setup -I have a Fedora 7 desktop with SER, mysql, and rtpproxy running. This is connected directly to the cable modem. There is not NAT in between. Firewall and SELinux are disabled.I have X-Lite client on my Laptop connected to a Wi-Fi router on a different IP. This router has NAT but the firewall is disabled in the router.On the far end, my sister has X-Lite on her desktop which is also behind NAT.Both X-Lite clients can REGISTER and the SIP server sends 200 OK. But can't make calls.When all clients and SER are on the same network with local IP addresses, I can make calls between the X-Lite clients and SJPhone on different computers. No problem. And Grandstream phones also work in the local setup.Any help is appreciated.HS

2 2

[OpenSER-Users] avp_substr, re-index's lists?
by Brandon Armstead 31 Aug '07

31 Aug '07

Hello, Say I have something like the following: $avp(a[0]) = "27*" $avp(a[1]) = "22*" Then I run avp_substr like the following: avp_subst("$avp(a)/g", "/\*/[0-9]/g"); As you would expect it to simply strip * from the avp list values, instead it decides to rewrite the index in numerical order, so the avp's would now look like: [ BAD RESULTS ] $avp(a[0]) = "22"; $avp(a[1]) = "27"; however the [ EXPECTED RESULTS ] $avp(a[0]) = "27"; $avp(a[1]) = "22"; Could someone please elaborate on why OpenSER would re-numerate the list index on a simple avp_substr, also any ideas on how I can get around this and stop this from happening? Thanks!

3 3

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