Hello List, Is there any setting that force asterisk to auto prune or forgot the peer information if for example x number of replies are not received
It keeps sending requests to the peer, I tried to turn off qualify and originating session timers to the peer but no luck Here is the message Reliably Transmitting (no NAT) to 10.200.1.55:5076: OPTIONS sip:[email protected]:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: "Unknown" <sip:[email protected]>;tag=as6c5371b0 To: <sip:[email protected]:5076;transport=tcp> Contact: <sip:[email protected]:5060;transport=TCP> Call-ID: [email protected]:5060 CSeq: 101 OPTIONS User-Agent: ASTPBX Date: Mon, 15 Apr 2013 15:25:09 GMT Session-Expires: 80 Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 --- [2013-04-15 11:25:09] WARNING[5183]: chan_sip.c:3386 __sip_xmit: sip_xmit of 0x7fad6c05c660 (len 609) to 10.200.1.55:5076 returned -2: Interrupted syste Before, when this retry was exceeded or connection was refused, asterisk restarted with the log message [2013-04-15 06:54:36] ERROR[5121] tcptls.c: Unable to connect SIP socket to 10.200.1.55:5075: Connection refused [2013-04-15 06:54:44] NOTICE[5167] loader.c: 2 modules will be loaded. I will produce a back trace later today and file a bug, I am using version 1.8.14.0 Please note, I have to stick with TCP because of packet loss in the network Any suggestions? Regards, Zohair Raza
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