Backtrace and logs attached here : https://issues.asterisk.org/jira/browse/ASTERISK-21447
Regards, Zohair Raza On Mon, Apr 15, 2013 at 11:13 PM, Mark Henry <[email protected]> wrote: > this is my secondary email > > Regards > Zohair > > > On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <[email protected]>wrote: > >> Tried disabling qualify and changing frequency with qualify=yes already, >> no luck :( >> >> >> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf < >> [email protected]> wrote: >> >>> I believe qualify parameters does help in doing so. Asterisk forgets >>> about the peer info when "qualify" are not acknowledged. You can also check >>> "qualifyfreq" to limit the number of qualifies for particular peer. >>> >>> >>> On Mon, Apr 15, 2013 at 7:37 AM, Zohair Raza < >>> [email protected]> wrote: >>> >>>> 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 >>>> >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>>> http://www.asterisk.org/hello >>>> >>>> asterisk-users mailing list >>>> To UNSUBSCRIBE or update options visit: >>>> http://lists.digium.com/mailman/listinfo/asterisk-users >>>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
