;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
<[email protected]>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 <[email protected]
> > 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 <
>> [email protected]> 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 <[email protected]>wrote:
>>>
>>>> this is my secondary email
>>>>
>>>> Regards
>>>> Zohair
>>>>
>>>>
>>>> On Mon, Apr 15, 2013 at 10:45 PM, Mark Henry <[email protected]>wrote:
>>>>
>>>>> Tried disabling qualify and changing frequency with qualify=yes
>>>>> already, no luck :(
>>>>>
>>>>>
>>>>> On Mon, Apr 15, 2013 at 10:11 PM, Mehroz Ashraf <
>>>>> [email protected]> 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 <
>>>>>> [email protected]> 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:[email protected]:5076;transport=tcp SIP/2.0
>>>>>>> Via: SIP/2.0/TCP 172.20.255.50:5060;branch=z9hG4bK0714eadd
>>>>>>> Max-Forwards: 70
>>>>>>> From: "Unknown" <sip:[email protected]>;tag=as6c5371b0
>>>>>>> To: <sip:[email protected]:5076;transport=tcp>
>>>>>>> Contact: <sip:[email protected]:5060;transport=TCP>
>>>>>>> Call-ID: [email protected]: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:
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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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>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
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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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>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
--
Bharat Lalcheta
--
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