SIP Agent configuration

[cavagnaro]

Hi guys,
I need some help here. I'm creating an Asterisk enviroment at my site, I already have a TServer A4400 running, agents declared and everything ok.
Now, using GAD when I try to make a call it says that the operation is not valid. Will check my GAD versión, maybe something to do with it.
But now, how do I create the Routing Point? I mean, on CME is already created, but how do I configure it on Asterisk (if needed) or how do I configure its annex section? Any help will be appreciated.

[cavagnaro]

Ok, now when I dial another extension what happens is that the sipphone of the agent who is making the call rings...I press answer and call drops...so I guess I'm doing something wrong. Do you have any configuration sample for your asterisk extensions?

[cavagnaro]

Ok now working, I can call but still calling the RP doesn't work...

any trick for associating the Agent with the RP??

Thanks

[cavagnaro]

Nobody? I thought this was used already...

[cavagnaro]

Worked! Just create the RP on the CME, no need to do anything on Asterisk side, except to send the calls to that number to SIP Server of Genesys. Now I have Alcatel sending calls to the SIP Server RP and asterisks agents answering hehe. Now will test OCS and more stuff.
My problem is that I was using an Alcatel agent and this was logging in using Alcatel loginID and not SIP Server loginID. Will test further as we want one single login for both SIP and Alcatel. If this works will be so cool.

[victor]

Hi, Cavagnaro,

actually we are trying to connect Genesys SIP T-Server to Asterix here, but having some setup difficulties. Can you put together some basic instructions on how to setup CME for Asterix? Something in the whitepaper from Genesys does not click right.

Best regards,
Vic

[cavagnaro]

Sure Vic! Will try do something more complete that Genesys white paper reaching URS even. Will post next week, I'm full right now...still can't answer Tony's even...

[victor]

Full? What are you eating there, in BRASIL!

[cavagnaro]

Peru! Peru! Damn! lol

[xember]

[quote]Ok, now when I dial another extension what happens is that the sipphone of the agent who is making the call rings...I press answer and call drops...so I guess I'm doing something wrong. Do you have any configuration sample for your asterisk extensions?[/quote]

Cavagnaro , how did you manage to get rid of the fact when calling an other extension the phone of the calling party starts to ring? I know it should be an option on the annex tap of the extension DN but for some reason I can not find out how.. any help is welcome on this..

I am using Counterpath Bria (latest) GDesktop (7.6 latest) SIP Server (7.6 latest)

Kind Regards,

Peter