[SR-Users] kamailio & vmware
Mack Hendricks
mack at dopensource.com
Wed Mar 28 15:20:16 CEST 2018
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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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
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