Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and not able to generate this scenario.
Regards, Bharat Lalcheta On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza <[email protected]>wrote: > 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 > -- Bharat Lalcheta
-- _____________________________________________________________________ -- 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
