Another Inbound Rout not working

Hi everybody. I understand this topic has been adressed many times, but after days of searching this and other forumsm I coukd find a clear answer. Probably because every installation has its own peculiarities, as well as the trunks with the providers.

I ask for help because I’m stuck, and when I say stuck I mean stuck for weeks. I still don’t get what’s missing.

First of all, I’m not expert with asterisk and freepbx, tho I have an average understanding of it.

My setup is as follow:

freepbx official distro installed in a VM (Parallels) for testing. INstall is ok as well as the first step of the installation. The system is online.

Asterisk is 11.4.0
freepbx is 2.11.0.0 according to the module admin

What I did after install is to set the sip trunk with my voip provider, and to set a sip phone for testing (internal number is 1000).

Using a softphone I logged into my server successfully, and I’m able to place calls to the telephone network, so I assume the outbound route is working well.

Problem is the inbound route.

If I define an inbound route leaving DID and CID blank, the call doesn’t hit my server, while if I enter my voip telephone number in the CID tab, it does. Freepbx status shows 1 active call.

Problem is that what I hear is the standard “The number you have dialed is not in service, …”.

This is the part I don’t understand. The call hits my server as far as I can tell, but I can’t understand how to route it to my SIP extension, even tho I set it in the “Set Destination” option (Inbound Route -> Set Destination -> Extensions -> 1000)

Here’s the log file:

[2013-08-29 13:08:51] VERBOSE[2152][C-0000000b] netsock2.c: == Using SIP RTP TOS bits 184 [2013-08-29 13:08:51] VERBOSE[2152][C-0000000b] netsock2.c: == Using SIP RTP CoS mark 5 [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:1] Set("SIP/from-trunk-0000000c", "GROUP()=OUT_2") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:2] Goto("SIP/from-trunk-0000000c", "from-trunk,41912200779,1") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Goto (from-trunk,41912200779,1) [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:1] Set("SIP/from-trunk-0000000c", "__FROM_DID=41912200779") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:2] NoOp("SIP/from-trunk-0000000c", "Received an unknown call with DID set to 41912200779") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:3] Goto("SIP/from-trunk-0000000c", "s,a2") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Goto (from-trunk,s,2) [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:2] Answer("SIP/from-trunk-0000000c", "") in new stack [2013-08-29 13:08:51] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:3] Wait("SIP/from-trunk-0000000c", "2") in new stack [2013-08-29 13:08:53] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:4] Playback("SIP/from-trunk-0000000c", "ss-noservice") in new stack [2013-08-29 13:08:53] VERBOSE[8571][C-0000000b] file.c: -- Playing 'ss-noservice.alaw' (language 'en') [2013-08-29 13:08:58] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:5] SayAlpha("SIP/from-trunk-0000000c", "41912200779") in new stack [2013-08-29 13:08:58] VERBOSE[8571][C-0000000b] file.c: -- Playing 'digits/4.alaw' (language 'en') [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: == Spawn extension (from-trunk, s, 5) exited non-zero on 'SIP/from-trunk-0000000c' [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:1] Macro("SIP/from-trunk-0000000c", "hangupcall,") in new stack [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/from-trunk-0000000c", "1?theend") in new stack [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: -- Goto (macro-hangupcall,s,3) [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:3] ExecIf("SIP/from-trunk-0000000c", "0?Set(CDR(recordingfile)=)") in new stack [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: -- Executing [[email protected]:4] Hangup("SIP/from-trunk-0000000c", "") in new stack [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/from-trunk-0000000c' in macro 'hangupcall' [2013-08-29 13:08:59] VERBOSE[8571][C-0000000b] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/from-trunk-0000000c' [2013-08-29 13:09:47] VERBOSE[2152][C-0000000c] netsock2.c: == Using SIP RTP TOS bits 184 [2013-08-29 13:09:47] VERBOSE[2152][C-0000000c] netsock2.c: == Using SIP RTP CoS mark 5 [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:1] Set("SIP/from-trunk-0000000d", "GROUP()=OUT_2") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:2] Goto("SIP/from-trunk-0000000d", "from-trunk,41912200779,1") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Goto (from-trunk,41912200779,1) [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:1] Set("SIP/from-trunk-0000000d", "__FROM_DID=41912200779") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:2] NoOp("SIP/from-trunk-0000000d", "Received an unknown call with DID set to 41912200779") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:3] Goto("SIP/from-trunk-0000000d", "s,a2") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Goto (from-trunk,s,2) [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:2] Answer("SIP/from-trunk-0000000d", "") in new stack [2013-08-29 13:09:47] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:3] Wait("SIP/from-trunk-0000000d", "2") in new stack [2013-08-29 13:09:49] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:4] Playback("SIP/from-trunk-0000000d", "ss-noservice") in new stack [2013-08-29 13:09:49] VERBOSE[8642][C-0000000c] file.c: -- Playing 'ss-noservice.alaw' (language 'en') [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: == Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/from-trunk-0000000d' [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:1] Macro("SIP/from-trunk-0000000d", "hangupcall,") in new stack [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/from-trunk-0000000d", "1?theend") in new stack [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: -- Goto (macro-hangupcall,s,3) [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:3] ExecIf("SIP/from-trunk-0000000d", "0?Set(CDR(recordingfile)=)") in new stack [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: -- Executing [[email protected]:4] Hangup("SIP/from-trunk-0000000d", "") in new stack [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/from-trunk-0000000d' in macro 'hangupcall' [2013-08-29 13:09:53] VERBOSE[8642][C-0000000c] pbx.c: == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/from-trunk-0000000d'

This is the part I can’t understand. Why can’t my setup route the call to the designated extension?

This is what I put in the trunk’s incoming setting, according to many hints I read in several forms, but I’m not sure if it’s useful:

host=sip.ticinocom.com type=peer nat=yes context=from-trunk

and the registration string:

user:[email protected]/41912200779

I can of course provide any useful information to resolve the matter.

Thank you for helping me.

Do you have an inbound route set for DID 41912200779 to your intended destination.

Please refer to: http://wiki.freepbx.org/display/F2/Inbound+Routes

Hi SkykingOH

well yes, I set it up according to the link you sent me, which is how I set it up anyway.

What is odd is that if I put the number in the DID Number field, the call doesn’t even hit my server, it just sounds busy.
If I put the number in the CallerID Number field, the call hits my server, but it doesn’t route to the extension.
If I leave both fields empty, the call doesn’t hit, sounds busy.
If I put the number in both fields, it hits the server but doesn’t route.

So I guess the number must be set at least in the CallerID number.

But still the call doesn’t route to my extension.