[asterisk-users] Asterisk SIP TCP

[Mark Henry]

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
>>
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- 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
>
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