[elchubi]

Hi, I followed instructions on voip-info to connect an Avaya Definity with an Asterisk box.
Avaya has all the channels "in-service/idle" so connection is working.
The thing is, when I dial the trunk access code at a telephone connected to the Avaya I get a dialing tone which is ok but then every key I pressed I got the error tone.

Asterisk log shows (for a key "5" pressed)

May 17 17:01:49 DEBUG[1381]: Manager received command 'Command'
May 17 17:02:21 VERBOSE[1381]: -- Starting simple switch on 'Zap/4-1'
May 17 17:02:21 VERBOSE[1381]: -- Accepting overlap call from '' to '' on channel 0/4, span 1
May 17 17:02:22 DEBUG[1381]: No such possible extension '5' in context 'from-pstn'
May 17 17:02:22 DEBUG[1381]: Set option AUDIO MODE, value: ON(1) on Zap/4-1
May 17 17:02:22 DEBUG[1381]: Hangup: channel: 4 index = 0, normal = 19, callwait = -1, thirdcall = -1
May 17 17:02:22 DEBUG[1381]: Not yet hungup... Calling hangup once with icause, and clearing call
May 17 17:02:22 DEBUG[1381]: Set option TDD MODE, value: OFF(0) on Zap/4-1
May 17 17:02:22 DEBUG[1381]: Updated conferencing on 4, with 0 conference users
May 17 17:02:22 DEBUG[1381]: Set option AUDIO MODE, value: OFF(0) on Zap/4-1
May 17 17:02:22 VERBOSE[1381]: -- Hungup 'Zap/4-1'

So asterisk is getting the "5" but not able to work with it.
- I want at the end is to give VoIP calls to my Avaya connected users.
- This is my first asterisk setup
- I'm using asterisk @ home 0.9 "vanilla", I just replaced zapata.conf and zaptel.conf configurations with the ones described at http://www.voip-info.org/wiki-Asterisk+Avaya

I would appreciate your help.
Thanks,

[elchubi]

just changed from "context=from-pstn" to "context=from-internal" and I'm able to dial into the pbx
Is this the appropiate context to work with?

As soon as I place the first VoIP call from an analog telephone connected to the Avaya I'll let you know

Regards,

[elchubi]

Ok, I set up a FWD account as an IAX2 Trunk, did the routing and I'm able to put 800 calls from any of the 500 phones (digital, analog, softphone and IP) at my site connected to the Avaya PBX.

Incoming calls are not needed, but just in case, I would appreciate if some one tells me how to do it.

Regards,

[elchubi]

Asterisk->VoIP is working

Avaya->Asterisk is working, so

Avaya->Asterisk->VoIP is working

VoIP->Asterisk is working

Asterisk->internal SIP is working, so

VoIP->Asterisk->internal SIP is working

I need help, please.

How can I get Asterisk ->Avaya calls

Regards,

[elchubi]

Ok, now VoIP-->Asterisk-->Avaya-->Avaya extension is working. It was an error of how I was defining the group.

Only one issue, I'm forwarding the VoIP call to the Avaya digital assistant who asks for a extension number. When doing the transfer Asterisk is hanging the call. I would appreciate your help.

Thanks,