[Users] Re: OpenSER + Asterisk - Music On Hold

[Carsten Bock]

Hi Edoardo,

Normally this would be handled by an Record-Route/Loose-Route construct.
When doing record-routing/loose-routing, the in-dialog request
"Re-INVITE" (for Music-On-Hold) should take the same route as the
initial request (following the route headers) and you should no longer
need to query the dispatcher-module for these in-dialog requests.
Maybe you could post your config, i guess then we could help a little
more.

Carsten


Am Montag, den 12.03.2007, 13:10 +0100 schrieb Edoardo Serra:
> Daniel,
> 	thanks for your interest in the problem.
> 
> I better analyzed the problem and found the point in it.
> I try to describe where I guess the problem is
> 
> When one of our users receive a call from the PSTN, the PSTN Gateway 
> (Asterisk) sends an INVITE at username at openser, the INVITE is correctly 
> forwarded to the user and the call is set up without problems.
> (RTP from PSTN gw to USER and SIP through OpenSER)
> 
> When the user wants to put the caller OnHold it sends an INVITE to 
> OpenSER but OpenSER forwards the INVITE to one of the PSTN GW using 
> dispatcher module.
> This way, if the INVITE is not forwarded to the PSTN GW which is 
> handling the call a second call is generated.
> 
> Do you have any suggestion ?
> Every kind of help is appreciated.
> 
> Sorry for not having sent a network capture, but is quite difficult to
> prepare such a capture on our system because it's always very busy
> 
> Hoping to hear from you soon
> 
> Regards
> 
> Edoardo
> 
> 
> Daniel-Constantin Mierla ha scritto:
> > Hello,
> > 
> > a network trace (ngrep or wireshark) will help to spot what might be the 
> > problem, otherwise is hard to guess.
> > 
> > Cheers,
> > Daniel
> > 
> > 
> > On 03/04/07 17:32, Edoardo Serra wrote:
> >> Hi all,
> >>     I have an OpenSER server in front of serveral Asterisk acting as a 
> >> load balancer and registrar server.
> >>
> >> We're offering both, inbound and outbound call services.
> >>
> >> When an outbound call is made, OpenSER, through the dispatcher module, 
> >> choose an Asterisk server to handle the media of the call.
> >>
> >> When an inbound call is received (by a PSTN GW interconnected with one 
> >> of the Asterisks), Asterisk calls SIP/username at openser.
> >>
> >> Media flows directly from user to Asterisks without using RTPProxy as 
> >> every Asterisk server has got a public IP Address..
> >>
> >> I have the following problem with MOH.
> >>
> >> If a user tries to put on hold an outbound call (placed by him) 
> >> everything is OK, Asterisk start playing MOH and stops when the user 
> >> wants to stop it.
> >>
> >> But, if a user wants to put on hold an inbound call (a call just 
> >> answered), as soon as it press the hold button another call to the 
> >> caller is originated and the first call is not put on hold by the 
> >> Asterisk
> >>
> >> I guess the problem is that, in this case, the asterisk doesn't 
> >> recognise the INVITE as a re-INVITE and originate a new call instead 
> >> of putting the other on hold.
> >>
> >> Do you have any idea on how to solve the problem ?
> >> Every suggestion is appreciated.
> >>
> >> Regards
> >>
> >> Edoardo Serra
> >>
> >>
> >> _______________________________________________
> >> Users mailing list
> >> Users at openser.org
> >> http://openser.org/cgi-bin/mailman/listinfo/users
> >>
> > 
> 
> 
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