29 Mar
2018
29 Mar
'18
1:09 a.m.
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(a)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(a)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(a)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 <http://dsiprouter.org/> - GUI focused on implementing
Kamailio to provide SIP Trunking and PBX Hosting Services
On Mar 27, 2018, at 6:06 PM, Alberto Llamas <albertollamaso(a)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(a)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
*"Internet is all about share"*
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