sr-users January 2007

sr-users@lists.kamailio.org

151 participants
232 discussions

by Shaun Hofer

Hi When ever MySQL Server (5.0.24a) on a remote machine is restarted, SER can't reconnect to the MySQL server. I see this problem has occurred before via mailing list: ERROR message : MySQL server has gone away (http://lists.iptel.org/pipermail/serusers/2006-April/027961.html) I am running debian package of SER 0.9.6-2 (ser-mysql-module 0.9.6-2) and MySQL Client 5.0.32 . I have tested that the MySQL client reconnects after server has been restarted. Is there something I'm missing that I'm still getting this problem ? Thanks --Shaun

17 years, 9 months

[Serusers] Problem with sl_send_reply: I won't send a reply for ACK.

by Ricardo Martinez

Hello list. I'm facing a problem with a UAC and i was hoping that someone can give me a hand here. I have a IP Phone calling to a PSTN number through SER and then a GW. 10.0.0.243 : IP Phone 10.0.0.246 : SER SIP Proxy 10.0.0.239 : GW SIP PSTN When the call is established and the "200 - OK" message arrives from the GW to the Proxy, the proxy re-route the message back to the Client, and finally the client respond with an ACK. Here is when te problem begins, i'm not sure if the ACK is the problem or maybe is a bug with my SER box. I'm using the Getting Started ser.cfg from iptel.org. You can see the debug here: U 10.0.0.239:5060 -> 10.0.0.246:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.246;branch=z9hG4bK34e7.0648f244.0. Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 INVITE. Supported: timer, replaces, early-session. User-Agent: A SIP Gateway. Contact: sip:005622408196@10.0.0.239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 247. Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>. . v=0. o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239. s=A Gateway SDP. c=IN IP4 10.0.0.239. t=1170173661 0. m=audio 23614 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.246:5060 -> 10.0.0.243:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.243:5060;rport=5060;branch=z9hG4bK407006395. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 INVITE. Supported: timer, replaces, early-session. User-Agent: A SIP Gateway. Contact: sip:005622408196@10.0.0.239. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO. Content-Type: application/sdp. Content-Length: 247. Record-Route: <sip:10.0.0.246;ftag=139103625;lr=on>. . v=0. o=005622408196 1170173661 1170173661 IN IP4 10.0.0.239. s=A Gateway SDP. c=IN IP4 10.0.0.239. t=1170173661 0. m=audio 23614 RTP/AVP 18 101. a=rtpmap:18 G729/8000/1. a=rtpmap:101 telephone-event/8000/1. a=fmtp:101 0-15. a=ptime:20. U 10.0.0.243:5060 -> 10.0.0.246:5060 ACK sip:0101005622408196@sipvoiss.desa.mydomain.net SIP/2.0. Via: SIP/2.0/UDP 10.0.0.243:5060;rport;branch=z9hG4bK1227697472. Route: <sip:10.0.0.246;ftag=139103625;lr=on>. From: <sip:5501234567@sipvoiss.desa.mydomain.net>;tag=139103625. To: <sip:0101005622408196@sipvoiss.desa.mydomain.net>;tag=d745f073a4. Call-ID: 90212623(a)10.0.0.243. CSeq: 21 ACK. Contact: <sip:5501234567@10.0.0.243:5060>. Max-Forwards: 70. User-Agent: S SIP User Agent / 1.10. Content-Length: 0. . Is this ACK ok?. The ACK hits the "Sanity Checks" (Max Forwards) and then breaks sending to console : "Warning: sl_send_reply: I won't send a reply for ACK!!" I was reading the RFC3261 because i'm not sure about the R-URI from this endpoint. Let me explain : RFC3261 : Section 12.1.2 The route set MUST be set to the list of URIs in the Record-Route header field from the response, taken in reverse order and preserving all URI parameters. If no Record-Route header field is present in the response, the route set MUST be set to the empty set. This route set, even if empty, overrides any pre-existing route set for future requests in this dialog. The remote target MUST be set to the URI from the Contact header field of the response. So, if this is correct the R-URI from the ACk must be set to sip:005622408196@10.0.0.239, and not the sip:0101005622408196@sipvoiss.desa.mydomain.net. Is this ok?. I think this is causing the ACK problem in my SER box. Can someone help me here? Thanks in advance. Best Regards, Ricardo Martinez.-

17 years, 9 months

[Serusers] a problem about radius and digest - openser patch

by TZieleniewski

_______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

17 years, 9 months

[Users] ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd)

by Hansi

Hello All, I have installed openser-1.1.1 with tls on my box. However, when I try to run openser. It returns me this error: ERROR: module version mismatch for /usr/local/lib/openser/modules/mysql.so; core: openser 1.1.1-tls (i386/freebsd); module: openser 1.1.1-notls (i386/freebsd). Would someone be kind enough to help me how to compile a mysql.so which supports tls? Thank you. "I'm a great believer in luck and I find the harder I work, the more I have of it." --------------------------------- No need to miss a message. Get email on-the-go with Yahoo! Mail for Mobile. Get started.

17 years, 9 months

[Serusers] Multiple Gateway Help

by Giovanni Chirico

Hi, How I can send calls to multiple Gateway? I would like to use the "dispatcher module". I made my configuration with dispatcher module, the calls are redirect to the gateway defined in the dispatcher.list file, but when I hung up the call the client always ring. (No BYE message). In attachment my ser.cfg file. Thanks in advance, John

17 years, 9 months

[Serusers] Getting 513 - Message too big for REGISTER message

by Sudhakar Patil

Hi, I am running into the attached errors. Appreciate your help. To add further clarification, looks like if (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); exit; }; max_len is getting converted to 0 (zero) and hence I keep getting 513 error message. I replaced max_len with numeric values but the effect is the same in that when comp_num in route.c is called the right value is always 0 (zero). Regards, Sudhakar. ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

17 years, 9 months

[Users] approaching testing phase for 1.2.0

by Daniel-Constantin Mierla

Hello everybody, we are approaching the testing phase for release 1.2.0. This week is planned to be the latest before cvs freezing. Then we will take care only of bug fixing, documentation update and small tunings. By this week we should be able to fix pending issues like script variables, dns failover and black lists, timers -- most of the code being already available on public cvs. If you have something else in mind, please send a mail to devel mailing list. Cheers, Daniel

17 years, 9 months

[Serusers] does ser support presence?

by Su Kerry

17 years, 9 months

[Serusers] Re: a problem about radius and digest

by TZieleniewski

17 years, 9 months

[Serusers] nathelper / rtpproxy

by Nhadie

Hi All, Can i use nathelper/rtpproxy on extension to extension and not use when an extension calls PSTN? Please see config if it makes sense. Thank You if (nat_uac_test("3")) { if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewritingn"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; if ((uri=~"^(sip:)?00[0-9]*@([a-z]+.)?mydomain.com") || (uri=~"^(sip:)?00[0-9]*@2.2.2.2")) { if (is_user_in("From", "onstun")) { if (isflagset(8)) {xlog("L_INFO", "THIS GROUP SHOULD NOT USE RTPPROXYn");}; # Strip first to digits (00) strip(2); prefix("37333#"); rewritehostport("10.10.10.10:5060"); consume_credentials(); t_relay_to_udp("10.10.10.10","5060"); break; } else { if (isflagset(8)) {xlog("L_INFO", "THIS WILL USE RTPPROXYn");}; if (isflagset(6)) route(1); <----- WILL CHECK SET FLAG 6 so that it will USE NATHELPER/RTPPROXY WILL THIS WORK? strip(2); prefix("37333#"); rewritehostport("10.10.10.10:5060"); consume_credentials(); t_relay_to_udp("10.10.10.10","5060"); break; } route[1] { xlog("L_INFO", "User-Agent behind NATn"); force_rtp_proxy(); if (method=="INVITE") { t_on_reply("1"); }; append_hf("P-Behind-NAT: Yesrn"); break; } ________________________________________________ Message sent using UebiMiau 2.7

17 years, 9 months

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sr-users January 2007