El Tuesday 10 July 2007 20:03:53 Christian Schlatter escribió:

Iñaki Baz Castillo wrote:

...

Hi, I'm starting using OpenSer with Asterisk.

In calls passing through Asterisk the caller or callee get music on hold of Asterisk when is put on hold. But of course, direct calls between OpenSer users don't get this music.

I'm thinking in the posibility of detecting "on hold" reinvites (maybe reading the "a=sendonly" of SDP body) and rewriting the SDP IP contact to a music on hold server.

You would also have to tell the music server where to send the rtp stream. And you'd have to keep dialog state on the proxy in order to capture the off-hold INVITE and tell the music server to stop sending media.

This all leads to a sip proxy acting as a B2BUA/3PCC server which needs a lot of resources and is difficult to implement. I don't say it is impossible but adding this functionality to openser would be quite a task.

Unfortunately even the IETF has no clear message on music on hold, no wonder sip phones normally do not support it in a standards compliant way. There is a draft called "SIP Service Examples" that already has 12 revisions and that includes an example on how music on hold should be implemented.

http://www1.tools.ietf.org/html/draft-ietf-sipping-service-examples-12#sect ion-2.3

Previous revisions of this draft followed a 3PCC model for music on hold (http://tech-invite.com/Ti-sip-service-3.html), whereas revision 12 now implements music on hold with a call transfer.

One can only hope that we will see products implementing these procedures.

Pingtel seems to offer a music-on-hold server for the 3PCC model: http://interop.pingtel.com/#moh

And SNOM phones seem to work fine together with the pingtel/sipX solution: http://sipx-wiki.calivia.com/index.php/HowTo_configure_SNOM_SIP_phone_with_ sipX#Music_On_Hold_.28MoH.29

Thanks a lot. I see that it's not an easy task. I'll try it whena have some tiem and if I get something decent I''ll tell it here.

Regards.

Maybe you have set report_ack ? http://www.openser.org/docs/modules/1.2.x/acc.html#AEN217

klaus

G.Jacobsen wrote:

I am using freeradius with openser 1.0.1.

My problem is that openser is emitting TWO radius start records: One for the INVITE with SIP method = 1 (INVITE) and also a second start record for SIP Method = 4 (ACK)

Is is normal that a start record is also emitted for the ACK ?

As long as the INVITE record is sent before the ACK record that is no real problem.

However every-now-and-then (no pattern observed) Openser sends the Radius start record with SIP Method = 4 (ACK) before the Record with SIP Method = 1 for the INVITE.

The problem is that the Radius record for the ACK does not contain the username from the credentials so that the username is not correctly set in the Radius record, instead of the correct username the callerid is used as the username.

How do I prevent that openser sends start records for the ACK ? Quite frankly I do not understand why a start record for ACK is emitted since I set the radius_flag only for INVITEs.

the relevant parts from my configuration:

modparam("acc", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 1) modparam("acc", "log_flag", 1) modparam("acc", "service_type", 15)

route{

..... # If this is an INVITE and not RE-INVITE start accounting if(is_method("INVITE") && !has_totag()){ # set the acc flags setflag(1); };

    if (is_method("ACK")) { #hop-by-hop ACKs are absorbed here
            t_newtran();
            # drop broken ACKs
            exit;
    };

Any hints are welcome.

TIA

Gerry

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