[SR-Users] kamailio & vmware

[Jean Cérien]

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 at 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 at 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 at 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 at 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 at 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.
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users at lists.kamailio.org
>>> 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
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
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