[asterisk-users] Asterisk SIP TCP
Bharat Lalcheta
bharatlalcheta at gmail.com
Tue Apr 16 01:12:59 CDT 2013
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their
registration expires, the
; information will _not_ be removed from
memory or the Asterisk database
; if you attempt to place a call to the
peer, the existing information
; will be used in spite of it having expired
;
; For realtime peers, when the peer is
retrieved from realtime storage,
; the registration information will be used
regardless of whether
; it has expired or not; if it expires
while the realtime peer
; is still in memory (due to caching or
other reasons), the
; information will not be removed from
realtime storage
Also remove all qualify related parameters and keepalive if set
Hope it will solve your problem
Regards,
Bharat Lalcheta
On Tue, Apr 16, 2013 at 11:26 AM, Zohair Raza
<engineerzuhairraza at gmail.com>wrote:
> 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--
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>>>>>>
>>>>>>
>>>>>> --
>>>>>> _____________________________________________________________________
>>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>>>
>>>>>
>>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>
>>
>>
>> --
>> 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
>>
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
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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
>
--
Bharat Lalcheta
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