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

[Brett Nemeroff]

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.
> --
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>
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