Asterisk As a SIP Redirect (for ex: TollByPass)


#1

I’m using an Asterisk box as a voicemail server, and TollByPass machine for a Tekelec telephony switch.

People call the Asterisk, get a dial tone, and can call some other number (wich isn’t local to them, but local to us using a bunch of PRI)

Let’s say CallerA calls our switch (t7000), wich sends the call to Asterisk,
Asterisk provides a dial tone (using DISA) and then calls the number the callerA entered using our switch as a peer…

Problem is the final call setup looks like this:
CallerA–>t7000–>Asterisk–>t7000–>Destination
I’d like to be able to get Asterisk out of the way when i know the destination… Maybe via a SIP 302 ?

Because as you see that call would use 4 channels on the switch when in fact only 2 should be needed:
CallerA–>t7000–>Destination

I red somewhere there might have been a patch for Asterisk (post 1.2) to do that, but i can’t find it.

If anyone got information on how to do that kind of thing, i’d be very interested!

Thank you all in advance


#2

Why not have the caller call the asterisk box directly and send the call out from there. Any specific reason whay you need the switch ?


#3

you could do it the otherway - have Asterisk be in front of the Nortel…

you would need (2) PRI cards in the asterisk… but you could do it like this:

PRI -> Asterisk -> Nortel

if someone calls in and does your app:

PRI -> Asterisk -> PRI


#4

Yes, lots of reasons, eventually we’ll have thousands of users, hundred’s of PRI, so Asterisk is not an option as a production class 5 switch…

[quote=“alohatone”]you could do it the otherway - have Asterisk be in front of the Nortel…
you would need (2) PRI cards in the asterisk… but you could do it like this:
PRI -> Asterisk -> Nortel
if someone calls in and does your app:
PRI -> Asterisk -> PRI[/quote]
Same as above, we curently got 3 DS3 (a ds3 is 28PRI…) coming in the switch (Tekelec, not Nortel) so having Asterisk in front is not an option…

We are thinking of trying to do this with a quad T1 card in Asterisk instead of purely SIP, but i’m pretty sure i’ll end up with the same behaviour…

What would be nice, is if i could send a SIP 302 redirect after the call is already established…
callerA calls asterisk, gets dial tone, enters the number to call, then asterisk would send SIP 302… but i’m pretty sure that’s not RFC compliant… I might do some test and try to do it anyway in asterisk with sipsak, or something similar…