Here is what I have, also attached sip show settings output and part of sip.conf in issues
[general] udpbindaddr=172.20.255.40 transport=udp,tcp tcpenable=yes tlsenable=no tcpbindaddr=172.20.255.40 directrtpsetup=no directmedia=yes allowguest=no match_auth_username=yes tos_sip=AF31 tos_audio=ef tos=0xB8 tos_video=af41 ; Sets TOS for RTP video packets. tos_text=af41 ; Sets TOS for RTP text packets. trustrpid = yes ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent (defaults to no) disallow=all allow=alaw allow=ulaw allow=g729 maxforwards=70 relaxdtmf=yes rpid_update = yes maxexpiry=400 minexpiry=60 defaultexpiry=300 qualify=yes ; notifycid = yes ; Control whether caller ID information is sent along with dialog-info+xml notifications (supported by snom phones) qualifyfreq=300 qualifypeers=1 qualifygap=2000 registertimeout=20 registerattempts=10 progressinband=never ignoreregexpire=yes On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta <[email protected]>wrote: > 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 >
-- _____________________________________________________________________ -- 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
