[asterisk-users] Asterisk SIP TCP

Tue Apr 16 00:56:49 CDT 2013

Here is what I have, also attached sip show settings output and part of
sip.conf in issues

[general]
udpbindaddr=172.20.255.40
transport=udp,tcp
tcpenable=yes
tlsenable=no
tcpbindaddr=172.20.255.40
directrtpsetup=no
directmedia=yes
allowguest=no
match_auth_username=yes
tos_sip=AF31
tos_audio=ef
tos=0xB8
tos_video=af41                 ; Sets TOS for RTP video packets.
tos_text=af41                  ; Sets TOS for RTP text packets.
trustrpid = yes                 ; If Remote-Party-ID should be trusted
sendrpid = yes                 ; If Remote-Party-ID should be sent
(defaults to no)
disallow=all
allow=alaw
allow=ulaw
allow=g729
maxforwards=70
relaxdtmf=yes
rpid_update = yes
maxexpiry=400
minexpiry=60
defaultexpiry=300
qualify=yes ;
notifycid = yes ; Control whether caller ID information is sent along with
dialog-info+xml notifications (supported by snom phones)
qualifyfreq=300
qualifypeers=1
qualifygap=2000
registertimeout=20
registerattempts=10
progressinband=never
ignoreregexpire=yes

On Tue, Apr 16, 2013 at 9:44 AM, Bharat Lalcheta
<bharatlalcheta at gmail.com>wrote:

> 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 --
>>>>>> 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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>>>>> To UNSUBSCRIBE or update options visit:
>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Bharat Lalcheta
>
> --
> _____________________________________________________________________
> -- 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
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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