[SR-Users] kamailio & vmware

[Joel Serrano]

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 at 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 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 - 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
>>>
>>>
>>>
>>> _______________________________________________
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>>> sr-users at lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>
>
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