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 <[email protected]> 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 <[email protected]> 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 <[email protected]> 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 <[email protected]> >> 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 <[email protected]> >> 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. >>> >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> [email protected] >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> >> -- >> Alberto Llamas >> Telecommunications Engineer >> dCAP | KPAC | SSCA >> >> >> >> *"Internet is all about share"* >> _______________________________________________ >> 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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