12 Mar
2007
12 Mar
'07
8:42 a.m.
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@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:
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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@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
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