[asterisk-users] SIP Connection Question

[Juan E. Rodríguez]

If * answers the call, it will be on the "loop" but with canreinvite or directrtp the media can be out of * and redirected to the final end point even if signaling goes through *.

For the trunk, you can have multiple simultaneous calls. I do not know about Mitel's licensing but with only one trunk you can have as much calls as * supports.

Saludos,
Juan E. Rodríguez


-----Original Message-----
From: "Dr. Kenneth Noisewater" <noisewaterphd at gmail.com>
Date: Thu, 01 Apr 2010 19:35:54 
To: <jerdguez at gmail.com>; Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] SIP Connection Question

On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
> Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302.
>
> If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you.
>
>
> Saludos,
> Juan E. Rodríguez
>
>
> -----Original Message-----
> From: Kenneth Noisewater<noisewaterphd at gmail.com>
> Date: Thu, 1 Apr 2010 16:50:47
> To:<asterisk-users at lists.digium.com>
> Subject: [asterisk-users] SIP Connection Question
>
>    
OK, so for instance if I passed a call to Asterisk and grabbed CID info 
and did some lookups and then transferred it back to mitel to route to a 
user, then * would be out of the call path (loop, whatever). But, if I 
were to answer that call in * with an IVR to collect caller input to use 
and then transferred the call back to the Mitel to route to the 
endpoint, * would remain in the call. Is that a correct understanding?

Also one more question, and please excuse my ignorance (I'm just a 
developer with pretty limited knowledge on the telephony side of things):

When I talk about connecting the Mitel box and the Asterisk box together 
via a SIP trunk, is that trunk equal to 1 analog line, or channel or 
whatever, or can I make as many connections as I want on that trunk? 
Again, my knowledge is a bit limited, and thusfar people have been using 
a lot of terms interchangably with me to add to my confusion :). This 
only concerns me because I'm pretty sure we have to buy a license for 
each SIP trunk with Mitel.

It would be really great if I could work out a solution like this, it 
will allow me to prove Asterisk's worth to my management, and open up a 
lot of doors for us and our internal apps. The Mitel SDK is 
unfortunately rather limited, but management is not in any way 
interested in jumping ship from Mitel to Asterisk. Personally, I say 
jump, I've had great experience with Asterisk, even in fairly heavy use 
situations. Anyone have any input on selling Asterisk to higher up's? I 
know there is the whole "enterprise support" aspect, but my team manages 
the Mitel stuff as it is anyway, and I think we'd all much rather be 
dealing with Asterisk/SER as the core solution.

Thanks everyone for your input!

Kenny