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

Daniel-Constantin Mierla miconda at gmail.com
Thu May 3 18:06:28 CEST 2018


Hello,

the problem is the UAC initiating the call -- the ACK must have in the
request URI the SIP address from Contact header of the 200ok, but the
ACK has the address pointing to Kamailio.

Moreover, to get the ACK and the rest of the requests part of the dialog
through Kamailio, you have to do record routing in kamailio ( see rr
module - e.g., record_route() and loose_route() functions as per default
config file use).

You need to adjust your sipp scenarios.

Cheers,
Daniel


On 03.05.18 17:42, Jeferson Oliveira wrote:
>
> 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 at gmail.com
>> <mailto: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 at servicescobrancas.com.br
>>     <mailto: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
>>         <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
>>         <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
>>>>             <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
>>>>             <https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing>
>>>>
>>>>
>>>>             Thank you very much.
>>>>
>>>>             -- 
>>>
>>>
>>>             _______________________________________________
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>>>             <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
>>>
>>
>>
>>         _______________________________________________
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>>
>>
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>>
>>
>
>
>
>
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-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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