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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd Max-Forwards: 70 From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 To: <sip:2271 at 10.200.1.55:5076;transport=tcp> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130415/9238d5f8/attachment.htm>
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 <engineerzuhairraza at gmail.com>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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd > Max-Forwards: 70 > From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 > To: <sip:2271 at 10.200.1.55:5076;transport=tcp> > Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> > Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130415/0b50faf5/attachment.htm>
Tried disabling qualify and changing frequency with qualify=yes already, no luck :( On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com>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 <engineerzuhairraza at gmail.com > > 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >> Max-Forwards: 70 >> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130415/225d71a9/attachment.htm>
this is my secondary email Regards Zohair On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430 at gmail.com> wrote:> Tried disabling qualify and changing frequency with qualify=yes already, > no luck :( > > > On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com > > 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 < >> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >>> Max-Forwards: 70 >>> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >>> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >>> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >>> 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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130415/2fd87839/attachment.htm>
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 <markhenry430 at gmail.com> wrote:> this is my secondary email > > Regards > Zohair > > > On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430 at gmail.com>wrote: > >> Tried disabling qualify and changing frequency with qualify=yes already, >> no luck :( >> >> >> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf < >> mehroz.ashraf85 at gmail.com> 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 < >>> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >>>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >>>> Max-Forwards: 70 >>>> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >>>> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >>>> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >>>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >>>> 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 >>> >> >> >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130416/18c0e90e/attachment.htm>
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 <engineerzuhairraza at gmail.com>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 <markhenry430 at gmail.com>wrote: > >> this is my secondary email >> >> Regards >> Zohair >> >> >> On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430 at gmail.com>wrote: >> >>> Tried disabling qualify and changing frequency with qualify=yes already, >>> no luck :( >>> >>> >>> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf < >>> mehroz.ashraf85 at gmail.com> 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 < >>>> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >>>>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >>>>> Max-Forwards: 70 >>>>> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >>>>> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >>>>> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >>>>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >>>>> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130416/d6c70a72/attachment.htm>
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 <bharatlalcheta at gmail.com>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 < > engineerzuhairraza at gmail.com> 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 <markhenry430 at gmail.com>wrote: >> >>> this is my secondary email >>> >>> Regards >>> Zohair >>> >>> >>> On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430 at gmail.com>wrote: >>> >>>> Tried disabling qualify and changing frequency with qualify=yes >>>> already, no luck :( >>>> >>>> >>>> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf < >>>> mehroz.ashraf85 at gmail.com> 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 < >>>>> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >>>>>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >>>>>> Max-Forwards: 70 >>>>>> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >>>>>> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >>>>>> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >>>>>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >>>>>> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130416/aab38041/attachment.htm>
;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
Also remove all qualify related parameters and keepalive if set
Hope it will solve your problem
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza
<engineerzuhairraza at gmail.com>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 <bharatlalcheta at
gmail.com
> > 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 <
>> engineerzuhairraza at gmail.com> 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 <markhenry430 at
gmail.com>wrote:
>>>
>>>> this is my secondary email
>>>>
>>>> Regards
>>>> Zohair
>>>>
>>>>
>>>> On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430
at gmail.com>wrote:
>>>>
>>>>> Tried disabling qualify and changing frequency with
qualify=yes
>>>>> already, no luck :(
>>>>>
>>>>>
>>>>> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf <
>>>>> mehroz.ashraf85 at gmail.com> 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 <
>>>>>> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp
SIP/2.0
>>>>>>> Via: SIP/2.0/TCP
172.20.255.50:5060;branch=z9hG4bK0714eadd
>>>>>>> Max-Forwards: 70
>>>>>>> From: "Unknown" <sip:Unknown at
172.20.255.50>;tag=as6c5371b0
>>>>>>> To: <sip:2271 at
10.200.1.55:5076;transport=tcp>
>>>>>>> Contact: <sip:Unknown at
172.20.255.50:5060;transport=TCP>
>>>>>>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at
172.20.255.50:5060
>>>>>>> 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
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On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta <bharatlalcheta at gmail.com>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 < > engineerzuhairraza at gmail.com> 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 < >> bharatlalcheta at gmail.com> 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 < >>> engineerzuhairraza at gmail.com> 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 <markhenry430 at gmail.com>wrote: >>>> >>>>> this is my secondary email >>>>> >>>>> Regards >>>>> Zohair >>>>> >>>>> >>>>> On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <markhenry430 at gmail.com>wrote: >>>>> >>>>>> Tried disabling qualify and changing frequency with qualify=yes >>>>>> already, no luck :( >>>>>> >>>>>> >>>>>> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf < >>>>>> mehroz.ashraf85 at gmail.com> 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 < >>>>>>> engineerzuhairraza at gmail.com> 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:2271 at 10.200.1.55:5076;transport=tcp SIP/2.0 >>>>>>>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd >>>>>>>> Max-Forwards: 70 >>>>>>>> From: "Unknown" <sip:Unknown at 172.20.255.50>;tag=as6c5371b0 >>>>>>>> To: <sip:2271 at 10.200.1.55:5076;transport=tcp> >>>>>>>> Contact: <sip:Unknown at 172.20.255.50:5060;transport=TCP> >>>>>>>> Call-ID: 433812eb21b0bb662afac65a129bb8b6 at 172.20.255.50:5060 >>>>>>>> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130416/98708d77/attachment.htm>