[asterisk-users] Asterisk SIP TCP

Zohair Raza engineerzuhairraza at gmail.com
Tue Apr 16 01:22:31 CDT 2013 

On Tue, Apr 16, 2013 at 10:12 AM, Bharat Lalcheta
<bharatlalcheta at gmail.com>wrote:

> ;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
>

I tried setting it to no already, but asterisk was keep trying to establish
connection at old ip and port


>  Also remove all qualify related parameters and keepalive if set
>
when qualify is set to no, does qualifyfreq have an effect? because I tried
qualify=no bu the qualifyfreq was set
at that time, I set qualifyfreq=300 but requests were going every few
seconds (around 30 secs)

One thing I doubt is Insecure field, it is set to no at the moment. By name
it is for security only but setting it insecure=port may effect?


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