[OpenSIPS-Users] 17 sec, recieve a bye and a hangup

[Peter den Hartog]

Thanks alot for you reply.

Asterisk is used because we have some agi stuff happening on incomming
calls. The sip trunk is registered on Asterisk. If i dial out, opensips uses
Asterisk because the extention is not in opensips (if i understand it
correctly) then Asterisk just uses his own sip trunk to dial outside. 

But for me it would be fine to use Opensips directly to make the connection
with the sip trunk, we  can leave asterisk out for now. 

1. There is two way audio, i can hear the other person talking, and he can
hear me 2.
2. no reinvite, i see a ok, and then a bye 
3. i don't know this yet, i can test it, i think i saw a empty ACK



Brett Nemeroff wrote:
> 
> I guess the question here is, what is asterisk doing for you? I personally
> would prefer the sip trunks right on opensips.. Asterisk is a kinda funny
> bottleneck in your architecture unless it's acting as some sort of media
> server (or TDM gateway).
> Some potential issues:
> 1. Do you have 2 way audio, some providers (gateways) will disco the call
> if
> there is one way audio for X seconds.
> 2. Do you see any reinvites happening? Some providers will re-invite calls
> after they are up and if the reinvite fails, it will tear down the call.
> 3. Where is the BYE coming from? Do you see any other signaling after the
> 200OK/ACKs you get? Do you see retransmissions of either the 200OK or ACK?
> If the signaling indicating the call was connected doesn't finish a proper
> ACK in both directions, the call will likely get hung up on.
> 
> 
> On Tue, Oct 6, 2009 at 8:17 AM, Peter den Hartog
> <peterdenhartog at gmail.com>wrote:
> 
>>
>> I'm trying to intergrate opensips with a allready running Asterisk
>> server.
>> The two servers are both on the same machine.
>>
>> I can recieve calls fine, Asterisk send them to my opensips installation,
>> and the opensips forwards the phone call to the right user. I can call
>> between the users on the network, with out any issue's so far so good.
>>
>> I have a sip trunk registered on Asterisk, and i use that for my in and
>> outgoing calls.
>>
>> But when i make an outside call, the call ends after 17 seconds. Looking
>> at
>> the sip messages i see that i recieve a bye, then the call is gone.
>>
>> Am i doing something wrong, should the sip trunk be directly in opensips?
>> and add that as a rewritehost? Or is this an Asterisk issue?
>>
>> My opensips is running on port 5090 (so are the phones) and my
>> asterisk+outside trunk is on 5060.
>> --
>> View this message in context:
>> http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html
>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> 

-- 
View this message in context: http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3775048.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.