NO JOY Just can't get outbound routes working

[Donald McMorris]

I have PIAF Green running Asterisk 11 and FreePBX 2.11 using CallCentric as a provider (already used them prior to hearing about Vitelity). I believe I have my trunk configuration set up correctly per other form posts and CallCentric's support web site, but I just can't make outgoing calls.

Trunk peer details:
context=from-trunk
fromdomain=callcentric.com
fromuser=1777NNNNNNN
host=callcentric.com
insecure=port,invite
secret=SECRET
type=peer
defaultuser=1777NNNNNNN
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw
Outbound route configuration is attached.
Made the following changes to Asterisk SIP Settings:
Default Context: from-pstn
SRV Lookup: Enabled
session-timers: refuse
Asterisk console shows this while attempting the call:
== Using SIP RTP CoS mark 5
-- Executing [17771234567@from-internal:1] ResetCDR("SIP/6473-00000026", "") in new stack
-- Executing [17771234567@from-internal:2] NoCDR("SIP/6473-00000026", "") in new stack
-- Executing [17771234567@from-internal:3] Progress("SIP/6473-00000026", "") in new stack
-- Executing [17771234567@from-internal:4] Wait("SIP/6473-00000026", "1") in new stack
> 0x7f68b0045790 -- Probation passed - setting RTP source address to 172.56.1.170:42707
-- Executing [17771234567@from-internal:5] Progress("SIP/6473-00000026", "") in new stack
-- Executing [17771234567@from-internal:6] Playback("SIP/6473-00000026", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/6473-00000026> Playing 'silence/1.gsm' (language 'en')
-- <SIP/6473-00000026> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
-- <SIP/6473-00000026> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [17771234567@from-internal:7] Wait("SIP/6473-00000026", "1") in new stack
-- Executing [17771234567@from-internal:8] Congestion("SIP/6473-00000026", "20") in new stack
== Spawn extension (from-internal, 17771234567, 8) exited non-zero on 'SIP/6473-00000026'
I tried clearing out the "context" of the extension, but the only change I noticed then was it read back the numbers I dialed...

 

[billsimon]

Looks like you forgot the username= field in your peer definition. It should be the same as your fromuser= field.

 

[pmosher441]

I wonder if there's more to it. I have the same trunk definition for Callcentric, i.e. no username field, and outgoing works fine.

 

[billsimon]

That's what I get for referring to the trunk definition on an Asterisk 1.4 server. Yes, the defaultuser= field should handle that part.

You might need to enable and paste more logging here so we can see what's going wrong.

 

[Donald McMorris]

What logs specifically would be helpful?

 

[rossiv]

What logs specifically would be helpful?

The call log you pasted above shows that the phone isn't hitting an outbound route. Asterisk doesn't know what do to because there's no matching dialplan.

Although, looking at what you dialed (assuming it's not sanitized), 17771234567 wouldn't be matched by your dialplan since the 1 in 123 isn't valid for N. N is 2-9. You'd need another line to match that.

 

[billsimon]

Nevermind! Hey, you fooled me. It is indeed your outbound route. You're dialing 17771234567 but your outbound route rules don't cover that pattern. The pattern you show is 1NXXNXXXXXX and the digit 1 (the second 1 in the string you are dialing) is not a member of N (2-9). (CallCentric is not obeying the NANP with their internal test number.) So if you want to dial that number, you'll need to add a pattern for it, such as "1777XXXXXXX".

 

[billsimon]

Nice catch rossiv. Sharper eye than mine.

 

[rossiv]

Nice catch rossiv. Sharper eye than mine.

Gotta love that timing!

 

[Donald McMorris]

Good catch indeed. Added a pattern for 1777XXXXXXX and it worked like a charm! Thanks very much!