Problem with outbound routes

Hi, I am very new to Asterisk.

Running Asterisk (Ver. 1.8.15.0)installed with FreePBX 2.10.1.1 Distro.

I set up a user and device. My softphone is registered with my box.

My SIP settings for my external and local networks are set up.

Codecs that are enabled are g729, ulaw, alaw, g726 and gsm. My service provider suggest g.729a. I don’t see it on the list.

Ok, next I did the Trunk settings. Put in my Trunk Name, which is my provider name and Outbound CallerID, which is the phone number they gave me. CID Options, allow any. Then my Trunk Name, which is my provider name. Then Peer Details are:

host=xxxxxxx.net
username=xxxxxxx
secret=xxxxxxx
type=peer

Then Outbound Routes. Route Name, my provider. Left everything else default and selected the trunk that I have configured.

My log file shows this when I dial from my softphone:

[2012-09-13 11:14:29] VERBOSE[3869] pbx.c: – Executing [[email protected]:1] ResetCDR(“SIP/102-00000001”, “”) in new stack
[2012-09-13 11:14:29] VERBOSE[3869] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/102-00000001”, “”) in new stack
[2012-09-13 11:14:29] VERBOSE[3869] pbx.c: – Executing [[email protected]:3] Progress(“SIP/102-00000001”, “”) in new stack
[2012-09-13 11:14:29] VERBOSE[3869] pbx.c: – Executing [[email protected]:4] Wait(“SIP/102-00000001”, “1”) in new stack
[2012-09-13 11:14:30] VERBOSE[3869] pbx.c: – Executing [[email protected]:5] Progress(“SIP/102-00000001”, “”) in new stack
[2012-09-13 11:14:30] VERBOSE[3869] pbx.c: – Executing [[email protected]:6] Playback(“SIP/102-00000001”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
[2012-09-13 11:14:30] VERBOSE[3869] file.c: – <SIP/102-00000001> Playing ‘silence/1.ulaw’ (language ‘en’)
[2012-09-13 11:14:31] VERBOSE[3869] file.c: – <SIP/102-00000001> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
[2012-09-13 11:14:33] VERBOSE[3869] file.c: – <SIP/102-00000001> Playing ‘check-number-dial-again.ulaw’ (language ‘en’)
[2012-09-13 11:14:36] VERBOSE[3869] pbx.c: – Executing [[email protected]:7] Wait(“SIP/102-00000001”, “1”) in new stack
[2012-09-13 11:14:37] VERBOSE[3869] pbx.c: – Executing [[email protected]:8] Congestion(“SIP/102-00000001”, “20”) in new stack
[2012-09-13 11:14:37] WARNING[3869] channel.c: Prodding channel ‘SIP/102-00000001’ failed
[2012-09-13 11:14:37] VERBOSE[3869] pbx.c: == Spawn extension (from-internal, 0117896399, 8) exited non-zero on ‘SIP/102-00000001’
[2012-09-13 11:14:37] VERBOSE[3869] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/102-00000001”, “”) in new stack
[2012-09-13 11:14:37] VERBOSE[3869] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/102-00000001’

Can someone please advise me?

Ok, I put 10 X in the Outbound Routes under Dial Patterns.

Now I get “all-circuits-busy-now&pls-try-call-later, noanswer”

That should be 011X.

If you are using g729 you will need licenses from Digium.

Thanks, I figured the g729 issue out, busy registering on Digium.

Thanks for the reply.

I’m still relatively new myself, but I do not see where it’s handing it off to anything. Can you post a screenshot of the screen where you have your out trunk (sans username & password) and your outbound route configured?

Cyber, thanks for your reply. I think my issue is the g729 codec which my provider uses. The did not inform me of it. I just did a search and found what my provider is using.

I am a lot further now.

I have been battling now for 2 weeks with this box. Understanding all the terminology etc.