[SR-Users] Fwd: Re: Kamailio with dispatcher not send BYE to Asterisk.
Amar Tinawi
amar.tinawi at gmail.com
Thu May 3 16:47:18 CEST 2018
Can you try tcp instead of udp ?
On Thu, May 3, 2018, 3:41 PM Jeferson Oliveira <
oliveira1.jeferson at 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 at 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/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing
>>
>>
>> Thank you very much.
>> --
>>
>>
>>
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>>
>
>
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