Many thanks for the help, I think I am getting closer, and obviously ;-) the issue is not caused by Kamailio !
under some load (still to confirm); when asterisk receives the relayed invite from K, it generates the reply message (100 trying) but this message does not reach the network, ie, I see the message in asterisk debug log, but not in the sngrep/pcap traces
Again, thanks for the help !
J.
On Wed, Mar 28, 2018 at 6:25 PM, Joel Serrano joel@gogii.net wrote:
Where are you t_relay()ing to?
Do you use dispatcher or similar?
Are you manually setting $du?
On Wed, Mar 28, 2018 at 3:09 PM, Jean Cérien cerien.jean@gmail.com wrote:
Thanks for the help
I've reproduced the issue on the test bed, with sipp to generate calls.
The issue appears in the second call - Asterisk places a call to Kamailio that should relay it to the carrier. Asterisk sends Invite, Kamailio replies with 100 and then nothing gets
out
of kamailio (I use sngrep on the box). I have traces in various routes in K, I see the call to t_relay, but I
see
nothing in sngrep - 2 or 4 secs later, K generates the 408
J.
On Wed, Mar 28, 2018 at 9:20 AM, Mack Hendricks mack@dopensource.com wrote:
Is the 200 getting back to the carrier? I’m assuming not. What does
the
INVITE and 200 message look like
On Mar 28, 2018, at 9:04 AM, Jean Cérien cerien.jean@gmail.com wrote:
Kamailio.
Here is the situation. Call arrives from voip provider to kamailio, it dispatches to asterisk, asterisk answers, and initiates another call
through
kamailio, and the voip provider.
K <-----------> Asterisk Invite -> <--- 100 <----180 <--- 200 <--- 200 retransmission,; happens 3-5 times Invite --> (same callid & cseq) <--- 200 retransmission,; happens 3-5 times
So, we see the asterisk dialplan has answered, and another call is
placed
form the asterisk K <-----------> Asterisk <------Invite 100 ----> (2 or 4 seconds later) 408 ---->
both nodes (kamailio and asterisk) show the same traces.
Any ideas would be greatly & truly appreciated, I am getting quite desperate about this one !
J.
On Wed, Mar 28, 2018 at 8:04 AM, Mack Hendricks ap@goflyball.com
wrote:
Are you getting the 408 from Asterisk or Kamailio? Perhaps you can provide a snippet of a sip capture.
Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com support: +888-907-2085 dSIPRouter - GUI focused on implementing Kamailio to provide SIP
Trunking
and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas albertollamaso@gmail.com wrote:
Hi Jean,
It might be something else. We do have an entire virtualized
environment
on Vmware with Asterisk, kamailios and another VoIP component without
any
issue with thousands of customers using it.
Regards,
On Tue, Mar 27, 2018 at 4:48 PM, Jean Cérien cerien.jean@gmail.com wrote:
Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I
actually
have no audio issues, but communication between the asterisk &
kamailio for
sip sometime fails - I get a few 408. I cant tell if this is network
related
or virtualisation related.
Anyone has advice on kamailio on a VM, when it only handles sip ?
Rgds J.
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-- Alberto Llamas Telecommunications Engineer dCAP | KPAC | SSCA
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