extension 's' rejected because extension not found

[kashutu]

Hi Guys,
Goautodial CE 2.0
VERSION: 2.2.1-237
BUILD: 100510-2015

I am facing this issue with Inbound calls. My outbound is working fine but somehow i am unable to figure out why my inbound calls are not getting through. My carrier is for Inboud as well as for outbound calls

Registering String:

register => xxxx:xxx@toronto.voip.ms:5060

Account Entry:

[voipms]
canreinvite=no
context=trunkinbound
host=toronto.voip.ms
secret=xxx
type=peer
username=xxx
disallow=all
allow=ulaw
allow=g729
fromuser=131171
trustrpid=yes

Global String:

TRUNK1=SIP/voipms

Dial Plan:

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(${VOIPTRUNK1}/${EXTEN:1},,Ttor)
exten => _91XXXXXXXXXX,3,Hangup

[quote]localhost*CLI> dial 917059992999@default
[Sep 24 09:01:10] WARNING[27199]: chan_oss.c:686 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory
[Sep 24 09:01:10] -- Executing [917059992999@default:1] AGI("Console/dsp", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 24 09:01:10] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 24 09:01:10] -- Executing [917059992999@default:2] Dial("Console/dsp", "SIP/voipms/17059992999||Ttor") in new stack
[Sep 24 09:01:10] -- Called voipms/17059992999
[Sep 24 09:01:12] -- SIP/voipms-0000002b is making progress passing it to Console/dsp
localhost*CLI> hangup
[Sep 24 09:01:16] == Spawn extension (default, 917059992999, 2) exited non-zero on 'Console/dsp'
[Sep 24 09:01:16] -- Executing [h@default:1] DeadAGI("Console/dsp", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[Sep 24 09:01:16] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 24 09:01:16] << Hangup on console >>
[Sep 24 09:01:33] NOTICE[3055]: chan_sip.c:15147 handle_request_invite: Call from '131171' to extension 's' rejected because extension not found.
[Sep 24 09:01:36] NOTICE[3055]: chan_sip.c:15147 handle_request_invite: Call from '131171' to extension 's' rejected because extension not found.
[Sep 24 09:01:37] NOTICE[3055]: chan_sip.c:15147 handle_request_invite: Call from '131171' to extension 's' rejected because extension not found.
quote]


Thank You for your help

[kashutu]

Ok this has been resolved. I was putting my USER ID for DID extension. I replaced it with my DID number and i am not getting the error anymore.

Thanks for looking anyways.

[spacejanitor]

I am having the exact same problem with the same carrier. I have tried Kashutu's solution but to no avail.

I've even tried changing the dialplan for my inbound carrier to:

Code: Select all

exten => s,1,AGI(agi://127.0.0.1:4577/call_log)
exten => s,2,Dial(${TRUNK1}/${EXTEN:1},,Ttor)
exten => s,3,Hangup

so that any "s" extensions are handled, but still I get the message:

Code: Select all

Call from 'carrierID' to extension 's' rejected because extension not found.

If anybody has any ideas on what to do here it would be greatly appreciated!

Thank you,

Mark

[kashutu]

Edit your extension.conf and put this under your [Inbound]

[trunkinbound]
; DID call routing process
;exten => _X.,1,AGI(agi-DID_route.agi)
exten => _'your inbound number here',1,AGI(agi-DID_route.agi)

After saving the changes, send reload command on asterisk CLI

[spacejanitor]

Thanks Kashutu...

Now I'm getting "ss no service" message "The number you have dialed is not in service", so I guess it is not triggering the DID route to my inbound group.

Here is my CLI output... strangely enough I do not see anywhere the actual DID number that I am dialing to reach this account. I DO see the number I am calling FROM to call the DID though (below changed to 4161234567).... any thoughts?

Code: Select all

<------------->
[Feb 10 09:40:22] --- (15 headers 14 lines) ---
[Feb 10 09:40:22] Sending to xxx.xxx.xxx.xxx : 5060 (NAT)
[Feb 10 09:40:22] Using INVITE request as basis request - 5d6f1f31093df6ab076eaef5714b8cfa@174.137.63.206
[Feb 10 09:40:22] Found peer 'voipms'
[Feb 10 09:40:22] Found RTP audio format 0
[Feb 10 09:40:22] Found RTP audio format 18
[Feb 10 09:40:22] Found RTP audio format 101
[Feb 10 09:40:22] Found audio description format PCMU for ID 0
[Feb 10 09:40:22] Found audio description format G729 for ID 18
[Feb 10 09:40:22] Found audio description format telephone-event for ID 101
[Feb 10 09:40:22] Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Feb 10 09:40:22] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Feb 10 09:40:22] Peer audio RTP is at port xxx.xxx.xxx.xxx:16402
[Feb 10 09:40:22] Looking for s in trunkinbound (domain 192.168.1.132)
[Feb 10 09:40:22] list_route: hop: <sip:4161234567@xxx.xxx.xxx.xxx>
[Feb 10 09:40:22]
<--- Transmitting (NAT) to xxx.xxx.xxx.xxx:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 4161234567:5060;branch=z9hG4bK47b4ec93;received=174.137.63.206;rport=5060
From: "ARROW PROFSSNL" <sip:4161234567@xxx.xxx.xxx.xxx>;tag=as2af3f2d4
To: <sip:s@192.168.1.132>
Call-ID: 5d6f1f31093df6ab076eaef5714b8cfa@xxx.xxx.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:s@192.168.1.132>
Content-Length: 0


<------------>
[Feb 10 09:40:22]     -- Executing [s@trunkinbound:1] AGI("SIP/voipms-000000a0", "agi-DID_route.agi") in new stack
[Feb 10 09:40:22]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Feb 10 09:40:22] ERROR[22198]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Feb 10 09:40:22]     -- AGI Script agi-DID_route.agi completed, returning 0
[Feb 10 09:40:22]     -- Executing [9998811112@default:1] Wait("SIP/voipms-000000a0", "2") in new stack