Help routing calls

Hi, I have a weird issue.

I have 2 SIP trunks from broadvoice and 1 from voicepulse.

I have them setup to go to different routes when calls come in. 2 go to IVR’s and 1 goes to an extension

When I call from “the real world” everything is fine. I can call anywhere dialing out from my system.

The problem is that I cannot call my own numbers from my system. It seems that Asterisk is grabbing the call and then playing the “we’re sorry, your call cannot be completed at this time” message.

Where/Why is this happening? What did I miss? Got to be something simple - please help - it’s driving me crazy.

all you inbound routes. Then add them one at a time and confirm operation each step of the way.

Other than that, which distro or hand compiled versions of what?

I am running Asterisk 1.4.21.2-1

I can call the Asterisk server from the real world without problems. It’s when the asterisk users try to dial one of the trunks.

For example:

Trunk 1 - 212-555-1234

Outbound route: 1NXXNXXXXXX -> Trunk 1

From an asterisk hosted sip phone, dial 1-212-555-1233 - works fine
Try dialing 1-212-555-1234 and you get the recording.

From the outside world ( my cell phone for example ) dial 1-212-555-1234 and I get the IVR as expected.

My inbound route is pretty simple - just point to either an extension or to the IVR

I am configuring everything using FreePBX

Something is not letting my call route out the sip trunk IF the call is to ANY of my sip trunks. It doesn’t matter if I try to call from trunk A to trunk B or trunk C or any combination.

I’m thinking it has something to do with these lines in the output of asterisk -r

– Executing [[email protected]:17] Macro(“SIP/1101-b7d17f48”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/1101-b7d17f48”, “0?bypass|1”) in new stack
– Executing [[email protected]:19] GotoIf(“SIP/1101-b7d17f48”, “0?customtrunk”) in new stack

Probably busy making the call. Try reversing the order of trunk in outbound.

1.4.21.2-1? I wasn’t aware of that one. Sounds like TB.

It is a new TB build and it’s been frustrating. Since all calls coming inbound from outside of our TB works fine it’s not critical, but it would be nice to test our own lines from our own phones…

If I force the call to go out trunk 1 while dialing the number for trunk 2 I get the same results. Each trunk has different outbound route - so for example dial 9 go out trunk 1, dial 8 go out trunk 2

I can even dial out from either of these trunks to a third trunk that comes in (with no outbound use) and get the same message.

Is there something that my box is seeing that is saying - hey, this is local so don’t dial out - and then fails? It’s not even handing the call to the trunk (I can look at the call details on the trunk at the service provider).

I’m almost ready to dump the TB build and go with another.

I’d try another provider…as a backup or additional trunk for testing. Vitelity is pretty inexpensive and worth a try to see if calls work with it. If it did, then something about the other provider would be suspect.

I can make calls out and receive calls fine. The issue is that I cannot seem to call my own trunks from my TB. If I term the trunks on a sip phone, I can call from one to the other. TB is not letting the call go out - it’s hijacking the call and playing the recording.

I am not convinced it is not the provider… or more to the point-how the provider is configured.