[asterisk-users] Asterisk SIP TCP

[Mark Henry]

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
>>>
>>>
>>> --
>>> _____________________________________________________________________
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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
>>
>> asterisk-users mailing list
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>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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