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

[Brett Nemeroff]

A trace of the whole call setup to hangup would be very helpful

On Tue, Oct 6, 2009 at 8:31 AM, Peter den Hartog
<peterdenhartog at gmail.com>wrote:

>
> 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.
> >>
> >> _______________________________________________
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> >> Users at lists.opensips.org
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >
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> >
> >
>
> --
> 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.
>
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