Using a sipgate DID with FreePBX

FreePBX Tips and Tricks
1

I’ve a FreePBX install and I’m going nuts. I must be missing something really, really simple.

I call my inbound number and I hear the “number is not in service” message.

The call is reaching my Asterisk server, as I can see it in the CLI.
Here is my trace:
[2013-08-13 14:39:45] VERBOSE[20986] asterisk.c: – Remote UNIX connection disconnected
[2013-08-13 14:40:07] VERBOSE[20957][C-00000029] netsock2.c: == Using SIP RTP TOS bits 184
[2013-08-13 14:40:07] VERBOSE[20957][C-00000029] netsock2.c: == Using SIP RTP CoS mark 5
[2013-08-13 14:40:07] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:1] NoOp(“SIP/SipgateTrk-0000002b”, “No DID or CID Match”) in new stack
[2013-08-13 14:40:07] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:2] Answer(“SIP/SipgateTrk-0000002b”, “”) in new stack
[2013-08-13 14:40:08] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:3] Wait(“SIP/SipgateTrk-0000002b”, “2”) in new stack
[2013-08-13 14:40:10] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:4] Playback(“SIP/SipgateTrk-0000002b”, “ss-noservice”) in new stack
[2013-08-13 14:40:10] VERBOSE[22603][C-00000029] file.c: – <SIP/SipgateTrk-0000002b> Playing ‘ss-noservice.gsm’ (language ‘en’)
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:5] SayAlpha(“SIP/SipgateTrk-0000002b”, “”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:6] Hangup(“SIP/SipgateTrk-0000002b”, “”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: == Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/SipgateTrk-0000002b’
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:1] Macro(“SIP/SipgateTrk-0000002b”, “hangupcall,”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/SipgateTrk-0000002b”, “1?theend”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/SipgateTrk-0000002b”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/SipgateTrk-0000002b”, “”) in new stack
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/SipgateTrk-0000002b’ in macro ‘hangupcall’
[2013-08-13 14:40:15] VERBOSE[22603][C-00000029] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/SipgateTrk-0000002b’
I’ve an inbound route set to ring an extension directly and no matter what I do to change the route, same thing. I get the ‘number not in service’ message.

I’ll post any information that I would need to - I really want to make this work. Sigh - I don’t want to edit the config files directly. I used to plain Asterisk and for some reason I feel like the web GUI is making me dumb(er).

I’m thinking that this:
pbx.c: – Executing [[email protected]-trunk:1] NoOp(“SIP/SipgateTrk-0000002b”, “No DID or CID Match”) in new stack may be the culprit of some sorts.

So, any takers? Anyone have the patience to give me some pointers?
Thanks - Glen

2

NoOp(“SIP/SipgateTrk-0000002b”, “No DID or CID Match”) in new stack

apparently you need to look closer at the sip headers as the normal request for a did is absent.

3

Okay - that’s why I posted. I’m clueless and asking for someone to point me in the right direction.
So, again, can someone give me a technically correct answer that will perhaps lead me to a solution?

"apparently you need to look closer at the sip headers as the normal request for a did is absent"

I don’t know what that means - can you please elaborate?

Thank you

4

sip set debug on

look for your did in there.

5

SIPGATE is not sending the DID where it is supposed to be. Do they not give Asterisk setup examples? It seems ridiculous you have to reverse engineer.

Why would you buy from someone that does not easily support the system when we have the Sipstation service that support the project?

6

I’ve been using sipgate for years for their excellently priced (free UK incoming local numbers). Never had any issues with calls not reaching me so this sounds like a configuration issue on your side not sipgate. I don’t mind SkykingOH’s advice to use sipstation, however that is a very US centric solution and if you are located elsewhere their offering is not really that attractive.

7

I will admit the US bias. My point was, if this is a new account why work with a carrier that doesn’t support Asterisk/FreePBX easily. There are so many, including SIPStation.

8