No Outbound -> Cannot be completed as dialed

So the version is 6.12.65-18.
Inbound is working fine and so does extension to extension. Im not using any dial patterns as I don’t think I really need them. Unless of course it’s a requirement which I’ve not been able to confirm I don’t want to use them.
We only have a few extensions and all numbers dialed are 1 + area code + 7-digit ##.

I use flowroute and my trunk setting are:
type=friend
secret=d0Nt933Kd0Nt933K
username=1231231
host=sip.flowroute.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
insecure=port,invite
fromdomain=sip.flowroute.com

Here’s the asterisk log after attempting an outbound call:

[2014-10-18 22:57:22] VERBOSE[2125][C-00000052] netsock2.c: == Using SIP RTP TOS bits 184 [2014-10-18 22:57:22] VERBOSE[2125][C-00000052] netsock2.c: == Using SIP RTP CoS mark 5 [2014-10-18 22:57:22] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:1] ResetCDR("SIP/3334-00000008", "") in new stack [2014-10-18 22:57:22] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:2] NoCDR("SIP/3334-00000008", "") in new stack [2014-10-18 22:57:22] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:3] Progress("SIP/3334-00000008", "") in new stack [2014-10-18 22:57:22] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:4] Wait("SIP/3334-00000008", "1") in new stack [2014-10-18 22:57:23] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:5] Progress("SIP/3334-00000008", "") in new stack [2014-10-18 22:57:23] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:6] Playback("SIP/3334-00000008", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack [2014-10-18 22:57:23] VERBOSE[12487][C-00000052] file.c: -- <SIP/3334-00000008> Playing 'silence/1.ulaw' (language 'en') [2014-10-18 22:57:24] VERBOSE[12487][C-00000052] file.c: -- <SIP/3334-00000008> Playing 'cannot-complete-as-dialed.ulaw' (language 'en') [2014-10-18 22:57:26] WARNING[12487][C-00000052] app_playback.c: Playback failed on SIP/3334-00000008 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer [2014-10-18 22:57:26] WARNING[12487][C-00000052] file.c: Failed to write frame [2014-10-18 22:57:26] VERBOSE[12487][C-00000052] file.c: -- <SIP/3334-00000008> Playing 'check-number-dial-again.ulaw' (language 'en') [2014-10-18 22:57:26] WARNING[12487][C-00000052] app_playback.c: Playback failed on SIP/3334-00000008 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer [2014-10-18 22:57:26] VERBOSE[12487][C-00000052] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/3334-00000008", "") in new stack [2014-10-18 22:57:26] VERBOSE[12487][C-00000052] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3334-00000008'

A few days into this now and I’d really appreciate some help.

Thanks a bunch,
John

yes, you need a matching 0utbound route and trunk

Forgot to mention that. Yes, I have a route setup. I guess that’s actually what I was referring to when I mentioned not using any dial pattern rules.

So my route Trunk Sequence for Matched Routes points to the trunk I above listed.
The route is also Not a part of any time group. There are no restrictions that I can tell or that I’ve made on the Outbound Route.

Thanks for Super incredibly quick response!

the trunk ’ above listed’ is an inbound trunk

It’s just beneath the

Outgoing Settings

section.

The registration string for Inbound is below.

well you might want to check the wiki and how your vsp wants it set . . . from-trunk is not realky approprite for outbound calls

That stack of outbound connection info is a direct copy from flowroutes faq. It’s the same as is on my other trixbox and works fine.

yet its not working. . .

that connection info works on an older machine which is running right now. On the new machine I have the same connection info and I get the message from the new machine “call cannot be completed as dialed”

sorry, all i can suggest is check your work again.

is it required that I setup Dial Patterns / a Dial Pattern?

Everything seems to be in order here. I really don’t know where to go from here.

Doesn’t the included asterisk log file tell you anything?

I really cant seem to find anything wrong. There is a pretty good chance though that if I were looking at something that was incorrect I wouldn’t know it.

Guess I’ll rest my eyeballs and prepare for another battle.

Thank you,
John

what’s ur dial pattern on ur trunk?

is it the same on the outbound route that’s using the trunk ?

I did not have any dial patterns setup. I mentioned this earlier in this thread. I also didn’t realize that there were two places to enter dial patterns.

I have set up a dial pattern for the trunk and for the route. Outbound calls work now!!!

Had no audio on outbound to cellphone for first call. Maybe i was trippin’ but the audio is there now.
Thank You!!!