[SR-Users] Fwd: Re: Kamailio with dispatcher not send BYE to Asterisk.

[Sergiu Pojoga]

As expected, the kamailio.pcap shows that in-dialog ACKs from sipp
10.110.7.242 are not being relayed to Asterisk 10.110.7.244 and so 10.110.7.244
keeps retransmitting.

On Thu, May 3, 2018 at 10:47 AM, Amar Tinawi <amar.tinawi at gmail.com> wrote:

> Can you try tcp instead of udp ?
>
> On Thu, May 3, 2018, 3:41 PM Jeferson Oliveira <oliveira1.jeferson@
> servicescobrancas.com.br> wrote:
>
>> Thank you for your reply Sergiu.
>>
>> Following url with the pcap capture of the two servers, kamailio.pcap and
>> asterisk.pcap.
>>
>> asterisk.pcap: https://drive.google.com/file/d/1DP--4NnSBsEQiN4n_
>> yfYoJzsmkv5fr1m/view?usp=sharing
>>
>> kamailio.pcap: https://drive.google.com/file/d/1agcA5Y3MuECdMl0mjnQ1dN4_
>> ht4YHDJU/view?usp=sharing
>>
>>
>> thanks a lot
>>
>> --
>>
>> On 05/02/2018 09:27 PM, Sergiu Pojoga wrote:
>>
>> 32 seconds is the default asterisk T2 timer. Probably some ACK is not
>> being relayed following BYE.
>>
>> Would help to see some sip traces.
>>
>> On Wed, May 2, 2018, 5:40 PM Jeferson Oliveira, <oliveira1.jeferson@
>> servicescobrancas.com.br> wrote:
>>
>>> Btw, the version of kamailio is 4.2.3.
>>>
>>> Thank you.
>>> --
>>> On 05/02/2018 06:29 PM, Jeferson Oliveira wrote:
>>>
>>> Hello everyone,
>>>
>>> I have an error that I have not yet been able to solve and would like
>>> the help of colleagues to indicate a correct path.
>>> The problem that is occurring is that when the client disconnects the
>>> call kamailio is not sending the BYE forward until arriving at the asterisk.
>>>
>>> Both in the test scenario and in the production scenario the problem is
>>> the same and the message I see in the capture is 404 Not here, msg this
>>> coming from kamailio.
>>>
>>> Production scenario.
>>>
>>> PSTN <----------> Dialer --------->kamailio -----------> asterisk1
>>>
>>> -----------> asterisk2
>>>
>>> Test scenario.
>>>
>>> sipp generated calls ------> kamailio -------> asterisk1
>>>
>>>                                                           ------->
>>> asterisk2
>>>
>>>
>>> When this occurs, the calls that are disconnected by the client are in a
>>> "zombie" state in asterisk, and end up being terminated by timeout with the
>>> following message in the asterisk CLI:
>>>
>>> *[Apr 25 17:49:59] WARNING[2121]: chan_sip.c:4072 retrans_pkt:
>>> Retransmission timeout reached on transmission 22-6073 at 10.110.7.242
>>> <22-6073 at 10.110.7.242> for seqno 1 (Critical Response) -- See
>>> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>>> <https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>*
>>> *Packet timed out after 31999ms with no response*
>>>
>>> In the sipp panel I see in the Retransmission column several
>>> incrementing counters, as per the attachment.
>>>
>>> If I take the kamailio from the move and point the sipp to only one of
>>> the asterisk, the retransmissions do not happen and BYE follows normally.
>>>
>>> My kamailio.cfg configuration file can be downloaded from this url:
>>> https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFj
>>> nNT/view?usp=sharing
>>>
>>>
>>> Thank you very much.
>>> --
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
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>>
>
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