Call Rejected because Extension not found

[kashutu]

Hi,

Goautodial CE 2.0
VERSION: 2.4-309a
BUILD: 110430-1642

Let me give you guys little back ground what have i done.

I have changed the SIP port 5060 to 8891 in sip.conf. Now i am able to register my softphone outside the network on port 8891. I have tried making outbound calls and it is working perfectly. However, when try to get and incoming call, here is the CLI error that i recieve:

[Oct 6 14:33:25] NOTICE[2897]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '7054811454' rejected because extension not found.
[Oct 6 14:33:27] NOTICE[2897]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '7054811454' rejected because extension not found.

Also i have tried SIP DEBUG and here is the output that i am getting:

<------------->
[Oct 6 14:23:53] --- (15 headers 14 lines) ---
[Oct 6 14:23:53] Sending to 174.137.63.206 : 58685 (NAT)
[Oct 6 14:23:53] Using INVITE request as basis request - 2cded5fd0df0677f073fbb0558b2ad49@174.137.63.206
[Oct 6 14:23:53] Found no matching peer or user for '174.137.63.206:58685'
[Oct 6 14:23:53] Found RTP audio format 0
[Oct 6 14:23:53] Found RTP audio format 18
[Oct 6 14:23:53] Found RTP audio format 101
[Oct 6 14:23:53] Found audio description format PCMU for ID 0
[Oct 6 14:23:53] Found audio description format G729 for ID 18
[Oct 6 14:23:53] Found audio description format telephone-event for ID 101
[Oct 6 14:23:53] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 6 14:23:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 6 14:23:53] Peer audio RTP is at port 174.137.63.206:10356
[Oct 6 14:23:53] Looking for 7054811454 in default (domain 173.248.228.98)
[Oct 6 14:23:53] LI>
<--- Reliably Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK44ae0f48;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as40ca57b3
To: <sip:7054811454@173.248.228.97:8891>;tag=as2a44ee0b
Call-ID: 2cded5fd0df0677f073fbb0558b2ad49@174.137.63.206
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 6 14:23:53] NOTICE[2897]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '7054811454' rejected because extension not found.
[Oct 6 14:23:53] Scheduling destruction of SIP dialog '2cded5fd0df0677f073fbb0558b2ad49@174.137.63.206' in 32000 ms (Method: INVITE)
[Oct 6 14:23:53] LI>
<--- SIP read from 174.137.63.206:58685 --->
ACK sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK44ae0f48;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as40ca57b3
To: <sip:7054811454@173.248.228.97:8891>;tag=as2a44ee0b
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 2cded5fd0df0677f073fbb0558b2ad49@174.137.63.206
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Content-Length: 0


<------------->
[Oct 6 14:23:53] --- (11 headers 0 lines) ---
[Oct 6 14:23:53] Really destroying SIP dialog '2cded5fd0df0677f073fbb0558b2ad49@174.137.63.206' Method: ACK
[Oct 6 14:23:54] LI>
<--- SIP read from 174.137.63.206:58685 --->
INVITE sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK09c08222;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as78422186
To: <sip:7054811454@173.248.228.97:8891>
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 0e0e90a50e1cb680465297e8259b046a@174.137.63.206
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Date: Thu, 06 Oct 2011 18:23:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 4665 4665 IN IP4 174.137.63.206
s=session
c=IN IP4 174.137.63.206
t=0 0
m=audio 12868 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Oct 6 14:23:54] --- (15 headers 14 lines) ---
[Oct 6 14:23:54] Sending to 174.137.63.206 : 58685 (NAT)
[Oct 6 14:23:54] Using INVITE request as basis request - 0e0e90a50e1cb680465297e8259b046a@174.137.63.206
[Oct 6 14:23:54] Found no matching peer or user for '174.137.63.206:58685'
[Oct 6 14:23:54] Found RTP audio format 0
[Oct 6 14:23:54] Found RTP audio format 18
[Oct 6 14:23:54] Found RTP audio format 101
[Oct 6 14:23:54] Found audio description format PCMU for ID 0
[Oct 6 14:23:54] Found audio description format G729 for ID 18
[Oct 6 14:23:54] Found audio description format telephone-event for ID 101
[Oct 6 14:23:54] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 6 14:23:54] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 6 14:23:54] Peer audio RTP is at port 174.137.63.206:12868
[Oct 6 14:23:54] Looking for 7054811454 in default (domain 173.248.228.98)
[Oct 6 14:23:54] LI>
<--- Reliably Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK09c08222;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as78422186
To: <sip:7054811454@173.248.228.97:8891>;tag=as07c123af
Call-ID: 0e0e90a50e1cb680465297e8259b046a@174.137.63.206
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 6 14:23:54] NOTICE[2897]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '7054811454' rejected because extension not found.
[Oct 6 14:23:54] Scheduling destruction of SIP dialog '0e0e90a50e1cb680465297e8259b046a@174.137.63.206' in 32000 ms (Method: INVITE)
[Oct 6 14:23:54] LI>
<--- SIP read from 174.137.63.206:58685 --->
ACK sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK09c08222;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as78422186
To: <sip:7054811454@173.248.228.97:8891>;tag=as07c123af
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 0e0e90a50e1cb680465297e8259b046a@174.137.63.206
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Content-Length: 0


<------------->
[Oct 6 14:23:54] --- (11 headers 0 lines) ---
[Oct 6 14:23:54] Really destroying SIP dialog '0e0e90a50e1cb680465297e8259b046a@174.137.63.206' Method: ACK
[Oct 6 14:23:56] LI>
<--- SIP read from 174.137.63.206:58685 --->
INVITE sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK0847188d;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7a116067
To: <sip:7054811454@173.248.228.97:8891>
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 08896128692f6ead261ecf067efed18a@174.137.63.206
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Date: Thu, 06 Oct 2011 18:23:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 4665 4665 IN IP4 174.137.63.206
s=session
c=IN IP4 174.137.63.206
t=0 0
m=audio 17576 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Oct 6 14:23:56] --- (15 headers 14 lines) ---
[Oct 6 14:23:56] Sending to 174.137.63.206 : 58685 (NAT)
[Oct 6 14:23:56] Using INVITE request as basis request - 08896128692f6ead261ecf067efed18a@174.137.63.206
[Oct 6 14:23:56] Found no matching peer or user for '174.137.63.206:58685'
[Oct 6 14:23:56] Found RTP audio format 0
[Oct 6 14:23:56] Found RTP audio format 18
[Oct 6 14:23:56] Found RTP audio format 101
[Oct 6 14:23:56] Found audio description format PCMU for ID 0
[Oct 6 14:23:56] Found audio description format G729 for ID 18
[Oct 6 14:23:56] Found audio description format telephone-event for ID 101
[Oct 6 14:23:56] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Oct 6 14:23:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 6 14:23:56] Peer audio RTP is at port 174.137.63.206:17576
[Oct 6 14:23:56] Looking for 7054811454 in default (domain 173.248.228.98)
[Oct 6 14:23:56] LI>
<--- Reliably Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK0847188d;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7a116067
To: <sip:7054811454@173.248.228.97:8891>;tag=as4cdb558f
Call-ID: 08896128692f6ead261ecf067efed18a@174.137.63.206
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 6 14:23:56] NOTICE[2897]: chan_sip.c:15147 handle_request_invite: Call from '' to extension '7054811454' rejected because extension not found.
[Oct 6 14:23:56] Scheduling destruction of SIP dialog '08896128692f6ead261ecf067efed18a@174.137.63.206' in 32000 ms (Method: INVITE)
[Oct 6 14:23:56] LI>
<--- SIP read from 174.137.63.206:58685 --->
ACK sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK0847188d;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7a116067
To: <sip:7054811454@173.248.228.97:8891>;tag=as4cdb558f
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 08896128692f6ead261ecf067efed18a@174.137.63.206
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Content-Length: 0


<------------->
[Oct 6 14:23:56] --- (11 headers 0 lines) ---
[Oct 6 14:23:56] Really destroying SIP dialog '08896128692f6ead261ecf067efed18a@174.137.63.206' Method: ACK
[Oct 6 14:23:58] LI>
<--- SIP read from 202.166.168.19:4069 --->

I have read alot of posts, and i am unable to figure out why am i not able to recieve calls. Please anyone help me with this problem.

I appreciate all your help

Thanks

[crisfurtado]

Call from '' to extension '7054811454' rejected because extension not found.

This is the error that is significant. It is telling you that asterisk does not know what to do with the call. To be precise your context is not programmed to handle this inbound call. Please configure a DID and route it to an ingroup or else agent or extension.

Please read though the Vicidial Manager manual and you will be able to set this up.[/quote]

[kashutu]

Ok, i have put this line in extensions.conf under

[default]

exten => _7054811454,1,AGI(agi-DID_route.agi)

Now i incoming is coming to vicidial however now when i call inbound number, i am getting the message please hold for the next availabe agent here is the CLI.

[Oct 8 04:05:12] -- Executing [7054811454@default:1] AGI("SIP/174.137.63.206-0000003e", "agi-DID_route.agi") in new stack
[Oct 8 04:05:12] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Oct 8 04:05:12] ERROR[32233]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Oct 8 04:05:12] -- AGI Script agi-DID_route.agi completed, returning 0
[Oct 8 04:05:12] -- Executing [99909*3***DID@default:1] Answer("SIP/174.137.63.206-0000003e", "") in new stack
[Oct 8 04:05:12] -- Executing [99909*3***DID@default:2] AGI("SIP/174.137.63.206-0000003e", "agi-VDAD_ALL_inbound.agi") in new stack
[Oct 8 04:05:12] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Oct 8 04:05:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:15] WARNING[32233]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:05:15] WARNING[32233]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:05:18] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:18] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:19] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:19] -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)
[Oct 8 04:05:20] -- Executing [h@default:1] DeadAGI("SIP/174.137.63.206-0000003e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 8 04:05:20] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

My DID ROUTE is set to IN_GROUP.

Here is the Debug report as well

<--- SIP read from 174.137.63.206:58685 --->
INVITE sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK3666ede4;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 102 INVITE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Date: Sat, 08 Oct 2011 08:11:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 475

v=0
o=root 4665 4665 IN IP4 174.137.63.206
s=session
c=IN IP4 174.137.63.206
t=0 0
m=audio 16486 RTP/AVP 0 3 8 112 5 10 7 18 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------->
[Oct 8 04:11:45] --- (15 headers 21 lines) ---
[Oct 8 04:11:45] Sending to 174.137.63.206 : 58685 (NAT)
[Oct 8 04:11:45] Using INVITE request as basis request - 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
[Oct 8 04:11:45] Found no matching peer or user for '174.137.63.206:58685'
[Oct 8 04:11:45] Found RTP audio format 0
[Oct 8 04:11:45] Found RTP audio format 3
[Oct 8 04:11:45] Found RTP audio format 8
[Oct 8 04:11:45] Found RTP audio format 112
[Oct 8 04:11:45] Found RTP audio format 5
[Oct 8 04:11:45] Found RTP audio format 10
[Oct 8 04:11:45] Found RTP audio format 7
[Oct 8 04:11:45] Found RTP audio format 18
[Oct 8 04:11:45] Found RTP audio format 111
[Oct 8 04:11:45] Found RTP audio format 101
[Oct 8 04:11:45] Found audio description format PCMU for ID 0
[Oct 8 04:11:45] Found audio description format GSM for ID 3
[Oct 8 04:11:45] Found audio description format PCMA for ID 8
[Oct 8 04:11:45] Found audio description format AAL2-G726-32 for ID 112
[Oct 8 04:11:45] Found audio description format DVI4 for ID 5
[Oct 8 04:11:45] Found audio description format L16 for ID 10
[Oct 8 04:11:45] Found audio description format LPC for ID 7
[Oct 8 04:11:45] Found audio description format G729 for ID 18
[Oct 8 04:11:45] Found audio description format G726-32 for ID 111
[Oct 8 04:11:45] Found audio description format telephone-event for ID 101
[Oct 8 04:11:45] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x9fe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|g726aal2)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[Oct 8 04:11:45] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 8 04:11:45] Peer audio RTP is at port 174.137.63.206:16486
[Oct 8 04:11:45] Looking for 7054811454 in default (domain 173.248.228.98)
[Oct 8 04:11:45] list_route: hop: <sip:0000123456@174.137.63.206:5060>
[Oct 8 04:11:45] LI>
<--- Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK3666ede4;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7054811454@173.248.228.98:8891>
Content-Length: 0


<------------>
[Oct 8 04:11:45] -- Executing [7054811454@default:1] AGI("SIP/174.137.63.206-0000003f", "agi-DID_route.agi") in new stack
[Oct 8 04:11:45] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Oct 8 04:11:45] ERROR[3485]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe
[Oct 8 04:11:45] -- AGI Script agi-DID_route.agi completed, returning 0
[Oct 8 04:11:45] -- Executing [99909*3***DID@default:1] Answer("SIP/174.137.63.206-0000003f", "") in new stack
[Oct 8 04:11:45] Audio is at 173.248.228.98 port 17418
[Oct 8 04:11:45] Adding codec 0x4 (ulaw) to SDP
[Oct 8 04:11:45] Adding codec 0x2 (gsm) to SDP
[Oct 8 04:11:45] Adding non-codec 0x1 (telephone-event) to SDP
[Oct 8 04:11:45] LI>
<--- Reliably Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK3666ede4;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>;tag=as528752de
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:7054811454@173.248.228.98:8891>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 2800 2800 IN IP4 173.248.228.98
s=session
c=IN IP4 173.248.228.98
t=0 0
m=audio 17418 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Oct 8 04:11:45] -- Executing [99909*3***DID@default:2] AGI("SIP/174.137.63.206-0000003f", "agi-VDAD_ALL_inbound.agi") in new stack
[Oct 8 04:11:45] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Oct 8 04:11:45] LI>
<--- SIP read from 174.137.63.206:58685 --->
ACK sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK25217cd7;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>;tag=as528752de
Contact: <sip:0000123456@174.137.63.206:5060>
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 102 ACK
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Content-Length: 0


<------------->
[Oct 8 04:11:45] --- (11 headers 0 lines) ---
[Oct 8 04:11:46] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:46] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:47] WARNING[3485]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:11:47] WARNING[3485]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:11:48] LI>
<--- SIP read from 117.102.40.6:48340 --->



<------------->
[Oct 8 04:11:51] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:51] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:51] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:52] -- Playing 'generic_hold' (escape_digits=) (sample_offset 0)
[Oct 8 04:11:58] WARNING[3485]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:11:58] WARNING[3485]: res_musiconhold.c:669 get_mohbyname: Music on Hold class 'default' not found
[Oct 8 04:11:59] LI>
<--- SIP read from 174.137.63.206:58685 --->
BYE sip:7054811454@173.248.228.98:8891 SIP/2.0
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK786bd8d0;rport
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>;tag=as528752de
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 103 BYE
User-Agent: VoIPMS/SERAST
Max-Forwards: 70
Remote-Party-ID: "0000123456" <sip:0000123456@174.137.63.206>;privacy=off;screen=no
Content-Length: 0


<------------->
[Oct 8 04:11:59] --- (10 headers 0 lines) ---
[Oct 8 04:11:59] Sending to 174.137.63.206 : 58685 (NAT)
[Oct 8 04:11:59] LI>
<--- Transmitting (NAT) to 174.137.63.206:58685 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 174.137.63.206:5060;branch=z9hG4bK786bd8d0;received=174.137.63.206;rport=58685
From: "0000123456" <sip:0000123456@174.137.63.206>;tag=as7b435757
To: <sip:7054811454@173.248.228.97:8891>;tag=as528752de
Call-ID: 3d10671d13d9a6f53c95da0441dcf002@174.137.63.206
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Oct 8 04:11:59] == Spawn extension (default, 99909*3***DID, 2) exited non-zero on 'SIP/174.137.63.206-0000003f'
[Oct 8 04:11:59] -- Executing [h@default:1] DeadAGI("SIP/174.137.63.206-0000003f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 8 04:11:59] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Oct 8 04:11:59] Really destroying SIP dialog '3d10671d13d9a6f53c95da0441dcf002@174.137.63.206' Method: BYE
[Oct 8 04:12:01] == Parsing '/etc/asterisk/manager.conf': [Oct 8 04:12:01] Found
[Oct 8 04:12:01] == Manager 'sendcron' logged on from 127.0.0.1
[Oct 8 04:12:01] == Manager 'sendcron' logged off from 127.0.0.1
[Oct 8 04:12:01] == Parsing '/etc/asterisk/manager.conf': [Oct 8 04:12:01] Found
[Oct 8 04:12:01] == Manager 'sendcron' logged on from 127.0.0.1

[kashutu]

I think i have figured out what was wrong. I had the DID Route set as IN_GROUP, but no agent was logged into any campagin, that is why whenever i used to call inbound number, i was getting a message wait for the available agent. I changed DID Route to Phone, mentioned the phone extension and logged into softphone and started getting calls.

Now my question is, what if i want to use the DID Route for PHONE and want to use multiple Phone Extensions to receive calls. How can i configure my DID Route to send the incoming calls on multiple Phone Extensions. I am not using any campaign, i just want to use this DID for incoming and outbound calls from softphone. I hope i am making sense. Please do ask me if i am not clear about the scenario.

Thanks