Avaya IP0 500 and Asterisk connected via SIP,Asterisk as SIP GATEWAY

[clarkstyx18]

So heres the scenario, I made sip trunk between IP office 500 and Asterisk Freepbx, SIP trunk is UP on both sides, Asterisk has a 3rd Party VOIP trunk , I need Avaya Phones to call international call using the VOIP provider which is configured in Asterisk. For Asterisk phones, I can route the calls through the VOIP provider without any issue but when it comes to the Avaya phones, its not going through.. I badly need your help guys.

 

[busster]

I have not done this using sip trunks, but have using a T-1 cable between IPO and Asterisk.

Is Asterisk setup to "dial 9" to get out? If so be sure that IPO is setup to pass the 9 and not remove it.

If you SSH into Asterisk or login on the Asterisk PC, go the CLI using asterisk -rvvvvv and watch as you make a call from IPO, you should get an idea of what is happening to the call.

 

[clarkstyx18]

For T1 connection, aside from the T1 cable, what other things are required for both avaya and asterisk? Im thinking this might be an alternative solution if I cant make it to work with SIP.



For dial plan on asterisk side, i made 00z. so calls will be routed to the voip provider.. for avaya side, I made short code 34N; , so from Avaya phone, Ill diall 3400440# , in this case Avaya will route calls to asterisk sip trunk, will send 00440 to asterisk so asterisk should choose the voip trunk and dial 440 to check the remaining of the voip account..



Ill try to run that command to see whats happening with the call.

 

[clarkstyx18]

@busster, heres the asterisk trace log

-- Executing [00440@from-trunk-sip-avaya ipo:1] Set("SIP/avaya ipo-0000003d", "GROUP()=OUT_2") in new stack
-- Executing [00440@from-trunk-sip-avaya ipo:2] Goto("SIP/avaya ipo-0000003d", "from-trunk|00440|1") in new stack
-- Goto (from-trunk,00440,1)
-- Executing [00440@from-trunk:1] Set("SIP/avaya ipo-0000003d", "__FROM_DID=00440") in new stack
-- Executing [00440@from-trunk:2] NoOp("SIP/avaya ipo-0000003d", "Received an unknown call with DID set to 00440") in new stack
-- Executing [00440@from-trunk:3] Goto("SIP/avaya ipo-0000003d", "s|a2") in new stack
-- Goto (from-trunk,s,2)
-- Executing [s@from-trunk:2] Answer("SIP/avaya ipo-0000003d", "") in new stack
-- Executing [s@from-trunk:3] Wait("SIP/avaya ipo-0000003d", "2") in new stack
-- Executing [s@from-trunk:4] Playback("SIP/avaya ipo-0000003d", "ss-noservice") in new stack
-- <SIP/avaya ipo-0000003d> Playing 'ss-noservice' (language 'en')
localhost*CLI>

 

[busster]

In Asterisk, do you have an outbound route 00XX. created? I am assuming that in IP Office, the user will be dialing 3400XXXXXXX, this will send 00XXXXXXX. If the 00 needs to be removed then the outbound route should be 00|XX. which will send all digits to which ever voip provider you have setup.

 

[busster]

A T-1 connection will essentially be the same setup. Place a T-1 cross over cable between the two and adjust your IPO and Asterisk routes.

 

[clarkstyx18]

@busster, yes, I already have 00z. as dial plan for the asterisk to route the calls to voip,

so you use the T1 cable to connect from Pri trunk of IPO to PRI trunk of Asterisk?

 

[busster]

Yes, but cross over, 1 to 4. 2 to 5.