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 [EMAIL PROTECTED], 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/[EMAIL PROTECTED] > >> > >> 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 > >> [email protected] > >> http://openser.org/cgi-bin/mailman/listinfo/users > >> > > > > > _______________________________________________ > Users mailing list > [email protected] > http://openser.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list [email protected] http://openser.org/cgi-bin/mailman/listinfo/users
