Exactly, do you have any tips on which parameter should I correct or add
in kamailio.cfg to solve this problem?
Thank you very much.
--
On 05/03/2018 12:01 PM, Sergiu Pojoga wrote:
> 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@gmail.com
>
mailto:amar.tinawi@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
>
mailto: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=s...
>
https://drive.google.com/file/d/1DP--4NnSBsEQiN4n_yfYoJzsmkv5fr1m/view?usp=sharing
>
> kamailio.pcap:
>
https://drive.google.com/file/d/1agcA5Y3MuECdMl0mjnQ1dN4_ht4YHDJU/view?usp=s...
>
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
>>
mailto: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@10.110.7.242
>>>
mailto:22-6073@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/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=s...
>>>
https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing
>>>
>>>
>>> Thank you very much.
>>>
>>> --
>>
>>
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