On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta <[email protected]>wrote:
> ;ignoreregexpire=yes ; Enabling this setting has two functions: > ; > ; For non-realtime peers, when their > registration expires, the > ; information will _not_ be removed from > memory or the Asterisk database > ; if you attempt to place a call to the > peer, the existing information > ; will be used in spite of it having > expired > ; > ; For realtime peers, when the peer is > retrieved from realtime storage, > ; the registration information will be > used regardless of whether > ; it has expired or not; if it expires > while the realtime peer > ; is still in memory (due to caching or > other reasons), the > ; information will not be removed from > realtime storage > I tried setting it to no already, but asterisk was keep trying to establish connection at old ip and port > Also remove all qualify related parameters and keepalive if set > when qualify is set to no, does qualifyfreq have an effect? because I tried qualify=no bu the qualifyfreq was set at that time, I set qualifyfreq=300 but requests were going every few seconds (around 30 secs) One thing I doubt is Insecure field, it is set to no at the moment. By name it is for security only but setting it insecure=port may effect? > > Hope it will solve your problem > > Regards, > > Bharat Lalcheta > > > On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza < > [email protected]> wrote: > >> 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 >> > > > > -- > 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 >
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