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 <[email protected]> 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 [email protected] >>> <[email protected]> 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 >>> [email protected] >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
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