[asterisk-users] Asterisk SIP TCP

[Bharat Lalcheta]

Can you give sip.conf ? I am using asterisk 1.8.15 on both udp and tcp and
not able to generate this scenario.

Regards,

Bharat Lalcheta



On Tue, Apr 16, 2013 at 11:03 AM, Zohair Raza
<engineerzuhairraza at gmail.com>wrote:

> 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
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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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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>>>
>>>
>>
>
> --
> _____________________________________________________________________
> -- 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
>



-- 
Bharat Lalcheta
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