Channel.c:6009 ast_request: No channel type registered for ” - Ast 11 - Inbound works - Outbound - “All Cir Busy

I am getting “All Circuits are Busy” on outbound calls, on a system that used to work.

Its CentOS 7 w/ Asterisk 11 and FreePBX 12.

Here are the errors from the debug:

[2017-01-03 14:50:54] WARNING[21663][C-00000003]: channel.c:6009 ast_request: No channel type registered for ''
[2017-01-03 14:50:54] WARNING[21663][C-00000003]: app_dial.c:2455 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
[2017-01-03 14:50:58] WARNING[21663][C-00000003]: channel.c:4861 ast_prod: Prodding channel 'SIP/2041-00000001' failed

The sip shows registered, and inbound calls work fine.

Any advice? I’m sure you’ll want more info, please let me know exactly what you need as I’m not sure what else to provide.

The dialplan is wrong. And you already knew that based on this:

You should double check that part of the dialplan.

I have triple checked all that I can think off. Here is the very simple dial plan that I have setup.

I removed and re-added the trunk, this is what registration has to say…

Host                                    dnsmgr Username       Refresh State                Reg.Time
sip.digiumcloud.net:5060                N      XYZ        120        Request Sent
1 SIP registrations.

I can’t tell if I’m registered or not, I think I am due to the bottom line, but the “state=Request Sent” makes me nervous and I thought it used to say “Registered” before I removed and re-added Trunk.

PS: I can still make inbound calls

Any help is appreciated, I’m fairly technical, but I need a bit of guidance how to debug this.

You need to check it from the bones. Enter in the asterisk cli and type:

diaplan show [email protected]

Replace XXXXXX with the dialed number. The output will show you the entire diaplan used to dial and you are looking for a DIAL command malformed. Paste the output here.

The registration has not received a reply. I don’t think the SIP traffic is making it back through your firewall.

May sure any SIP ALG’s are turned off. For a test forward UDP 5060 to the server and see if that caused the state to go to registered.

1 Like
node1*CLI> dialplan show [email protected]
[ Included context 'outrt-1' created by 'pbx_config' ]
  '_91NXXNXXXXXX' => 1. Macro(user-callerid,LIMIT,EXTERNAL,)       [pbx_config]
                2. Gosub(sub-record-check,s,1(out,${EXTEN},dontcare)) [pbx_config]
                3. ExecIf($[ "${CALLEE_ACCOUNCODE}" != "" ] ?Set(CDR(accountcode)=${CALLEE_ACCOUNCODE})) [pbx_config]
                4. Set(MOHCLASS=${IF($["${MOHCLASS}"=""]?default:${MOHCLASS})}) [pbx_config]
                5. ExecIf($["${KEEPCID}"!="TRUE" & ${LEN(${DB(AMPUSER/${AMPUSER}/outboundcid)})}=0 & ${LEN(${TRUNKCIDOVERRIDE})}=0]?Set(TRUNKCIDOVERRIDE=2155554567)) [pbx_config]
                6. Set(_NODEST=)                              [pbx_config]
                7. Macro(dialout-trunk,4#${EXTEN:1},#off)     [pbx_config]
                8. Macro(outisbusy,)                          [pbx_config]

[ Included context 'bad-number' created by 'pbx_config' ]
  '_X.' =>          1. ResetCDR()                                 [pbx_config]
                2. NoCDR()                                    [pbx_config]
                3. Progress()                                 [pbx_config]
                4. Wait(1)                                    [pbx_config]
                5. Progress()                                 [pbx_config]
                6. Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer) [pbx_config]
                7. Wait(1)                                    [pbx_config]
                8. Congestion(20)                             [pbx_config]
                9. Hangup()                                   [pbx_config]

-= 2 extensions (17 priorities) in 2 contexts. =-
[[email protected] ~]# asterisk -rx 'sip show registry'
Host                                    dnsmgr Username       Refresh State                Reg.Time
sip.digiumcloud.net:5060                N      USERNAME       105 Registered           Wed, 04 Jan 2017 17:15:45
1 SIP registrations.

You are dialing 91123456789 but your “outbound route” has 91NXXNXXXXXX which is 1 digit more than the dialed one, that’s why that number enter in the “bad-number” context.

Dial a number with the same length of your Outbound route.

I actually just wrote that in manually, and removed my Cell Number, I’m pretty sure I tried all variatons of w and w/o #1, I think the issue was w/ my that on my Outbound Routes it wasn’t using the correct trunk. Regardless, I have started fresh and its working now. Thank you for your suggestions, I’m always learning via troubleshooting/breaking things :slight_smile: