TIPS Multiple GV Trunks - Multpile Extensions - 1:1 inbound & outbound calling - Help please

[_steve_]

I have tried both ways.. 1NXXNXXXXXX and NXXNXXXXXX

Each time, I am presented with the lovely "your call cannot be completed as dialed..."

 

[_steve_]

To answer your question re: calling one sip phone from another, I have been able to make inbound calls (from a third google voice account (not attached to the system in any way). I can call each of the installed trunks and they are directed to the appropriate extensions.

I am also able to dial from one extension to another properly.

 

[TwigsUSAN]

Try dialing 11+10 digits.

 

[_steve_]

11+10? which 11 and which 10?

 

[TwigsUSAN]

Dial 11 and a ten digit phone number.

So 113125551212 as an example.

 

[_steve_]

ok, I have tried:
9+1NXXNXXXXXX
1+1NXXNXXXXXX

Still nothing.

 

[_steve_]

root@pbx:/etc/asterisk $ tailf /var/log/asterisk/full
[2013-12-27 16:29:10] VERBOSE[12122][C-00000007] netsock2.c: == Using SIP RTP TOS bits 184
[2013-12-27 16:29:10] VERBOSE[12122][C-00000007] netsock2.c: == Using SIP RTP CoS mark 5
[2013-12-27 16:29:10] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:1] ResetCDR("SIP/<sanitized>-00000007", "") in new stack
[2013-12-27 16:29:10] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:2] NoCDR("SIP/<sanitized>-00000007", "") in new stack
[2013-12-27 16:29:10] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:3] Progress("SIP/<sanitized>-00000007", "") in new stack
[2013-12-27 16:29:10] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:4] Wait("SIP/<sanitized>-00000007", "1") in new stack
[2013-12-27 16:29:11] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:5] Progress("SIP/<sanitized>-00000007", "") in new stack
[2013-12-27 16:29:11] VERBOSE[13924][C-00000007] pbx.c: -- Executing [11XXXXXXXXXX@Local:6] Playback("SIP/<sanitized>-00000007", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2013-12-27 16:29:11] VERBOSE[13924][C-00000007] file.c: -- <SIP/<sanitized>-00000007> Playing 'silence/1.gsm' (language 'en')
[2013-12-27 16:29:12] VERBOSE[13924][C-00000007] file.c: -- <SIP/<sanitized>-00000007> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: == Spawn extension (Local, 11XXXXXXXXXX, 6) exited non-zero on 'SIP/<sanitized>-00000007'
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: -- Executing [h@Local:1] Macro("SIP/<sanitized>-00000007", "hangupcall,") in new stack
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/<sanitized>-00000007", "1?theend") in new stack
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: -- Goto (macro-hangupcall,s,3)
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf("SIP/<sanitized>-00000007", "0?Set(CDR(recordingfile)=)") in new stack
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup("SIP/<sanitized>-00000007", "") in new stack
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/sjb1963-denver-00000007' in macro 'hangupcall'
[2013-12-27 16:29:14] VERBOSE[13924][C-00000007] pbx.c: == Spawn extension (Local, h, 1) exited non-zero on 'SIP/<sanitized>-00000007'

 

[TwigsUSAN]

9 won't work if you don't have a dial plan setup. Try deleting the dial plans and start over. If it still doesn't work, the. You would have to give us the message logs when you make a call.

 

[_steve_]

are you talking about the dialing plan on the extensions.conf file? Or the dialing plan on the screen shots I posted?

 

[TwigsUSAN]

The screen shots. You shouldn't be modifying anything under the extensions.conf.

 

[_steve_]

Ok, well, I think I'm going to have to start over.
Editing those files (extensions.conf & sip.conf) are the ONLY way I have gotten anything to work at all.
More than just a little frustrated. It sure would be nice if 98% of all the crap in the web didn't have a freaking date in it to be able to tell what was pertinent and what wasn't.

 

[TwigsUSAN]

Re-image and start over.

Get your base install up and running. Get phone to phone calls working. Get inbound GV calls working. Then we can tackle the outbound routes.

 

[_steve_]

that's where I am right now.
I can do everything but dial out

 

[TwigsUSAN]

But with a fresh start? Who know what might have been broken with modifying different files.

 

[_steve_]

Well, I do appreciate your patience. I am not new to IT or networking, but Telephony has always been one of those things that makes my head explode. Now I'm biting the bullet and trying to learn this stuff. I haven't been out from in front of the computer in several days now and I'm more than just a little burnt.

So, I am taking the PIAF-Green-20650 distro I have and installing it on Virtual Box again. I didn't destroy the old one, but I won't be going back to it any time soon I'm guessing.

I'll let you know when I get back to a starting place. Thanks again for all your help.

 

[TwigsUSAN]

Are you using an ISO or grabbing the Virtualbox file?

 

[_steve_]

I'm importing the ova into virtualbox. I have it up now, nothing updated yet and changing base passwords... but it's a lot quicker.

 

[_steve_]

I STILL don't have the license key for that freaking free software yet. I'm going to work without it. This is ridiculous.

 

[TwigsUSAN]

They are closed for the Holidays and probably haven't approved it yet.

 

[_steve_]

Ok. I'm back. I got the base install done. I created one single extension and installed one single google voice account.
I selected "Asterisk Sip Settings" and told it to "auto Configure"
I set the extension name and password.

Unable to register with my phone (iPhone using ground wire - worked before), or with the Yate softphone.
This is where I started editing the sip.conf before.

Ideas?