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 >
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