Hi guys, Am having a strange SIP problem in my call centre. The call centre has about 70 SIP agents (some of the are using SIP hard phones, other SIP softphones), and occasionally most of the SIP peers (hardphones and softphones) become UNREACHABLE and then after few second again REACHABLE. Some hardphones and softphones work perfectly normal during that period (even normally responding to OPTIONS message), but most of them get UNREACHABLE.
I don't have NAT - phones and Asterisk are in the same subnet, so nothing complicated really (regarding network configuration). I'm currently suspecting my network to be the problem, but I would just like to confirm with you guys, if you have any similar experiences, what could be causing this? Please, see bellow one of the sample SIP traces. Regards, Alex Jan 1 11:17:42 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: "asterisk" <sip:[email protected]>;tag=as02e1afaa To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:45 VERBOSE[6046] logger.c: Retransmitting #1 (no NAT) to 165.11.1.41:5060: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=a4bG4bK4b7hf375;rport From: "asterisk" <sip:[email protected]>;tag=as02e1afaa To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:46 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now UNREACHABLE! Last qualify: 14 Jan 1 11:17:56 VERBOSE[6046] logger.c: Reliably Transmitting (no NAT) to 165.11.1.41:5060: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: "asterisk" <sip:[email protected]>;tag=as796f6356 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 01 Jan 2010 11:17:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: <-- SIP read from 165.11.1.41:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 165.11.1.50:5060;branch=z2h16b637dKh2fd;rport From: "asterisk" <sip:[email protected]>;tag=as796f6356 To: <sip:[email protected]>;tag=5A4BF5F8-460290A9 CSeq: 102 OPTIONS Call-ID: [email protected] Contact: <sip:[email protected]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.2.0.0047 Content-Length: 0 Jan 1 11:17:56 VERBOSE[6046] logger.c: --- (10 headers 0 lines) --- Jan 1 11:17:56 NOTICE[6046] chan_sip.c: Peer 'TestPhone1' is now REACHABLE! (16ms / 10000ms) _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
