[asterisk-users] Asterisk SIP TCP

[Zohair Raza]

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