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.
--
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@gmail.com> 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 22-6073@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/ 1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFj nNT/view?usp=sharing
Thank you very much.
--
_________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr- users
_________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr- users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr- users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users