[asterisk-users] Asterisk SIP TCP

[Zohair Raza]

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>