Can you try tcp instead of udp ? On Thu, May 3, 2018, 3:41 PM Jeferson Oliveira < [email protected]> 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, < > [email protected]> 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/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/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 >
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