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 <[email protected]
<mailto:[email protected]>> wrote:
Can you try tcp instead of udp ?
On Thu, May 3, 2018, 3:41 PM Jeferson Oliveira
<[email protected]
<mailto:[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
<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,
<[email protected]
<mailto:[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]
<mailto:[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
<https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=sharing>
Thank you very much.
--
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