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

[Jeferson Oliveira]

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 
> <mailto: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 <mailto:22-6073 at 10.110.7.242> for seqno 1
>>     (Critical Response) -- See
>>     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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