Inbound call rejected

[JCSANHUEZA]

Hello

I just pointed a DID to an inbound campaign with its corresponding inbound group, but I getting the following error message at CLI

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[Jul  8 19:57:21]   == Parsing '/etc/asterisk/manager.conf': [Jul  8 19:57:21] Found
[Jul  8 19:57:21]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  8 19:57:30]        > Channel SIP/200-04942a80 was answered.
[Jul  8 19:57:30]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  8 19:57:30]     -- Executing [8600051@default:1] MeetMe("SIP/200-04942a80", "8600051|F") in new stack
[Jul  8 19:57:30]   == Parsing '/etc/asterisk/meetme.conf': [Jul  8 19:57:30] Found
[Jul  8 19:57:30]   == Parsing '/etc/asterisk/meetme-vicidial.conf': [Jul  8 19:57:30] Found
[Jul  8 19:57:30]     -- Created MeetMe conference 1023 for conference '8600051'
[Jul  8 19:57:30]     -- <SIP/200-04942a80> Playing 'conf-onlyperson' (language 'en')

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[Jul  8 19:59:18] NOTICE[2944]: chan_sip.c:14035 handle_request_invite: Call from '' to extension 'XXXXXXXXX'' rejected because extension not found.

SIPTRUNK

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[sipcom]
username=123456789
user=123456789
type=peer
secret=987654321
host=sip.
fromusername=123456789
fromuser=123456789
fromdomain=sip.sipcom.com
disallow=all
allow=g729
allow=alaw
allow=ulaw
nat=yes
context=trunkinbound

extensions.conf

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[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

DID

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Active: Y
DID Extenision:123456789
DID Route: IN_GROUP
In-Group ID: SALES
In-Group Call Handle Method: CID
In-Group Agent Search Method: LB
In-Group Phone Code: 1

Remaining fields were set with default values

In-group

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Group ID: SALES
Call Time: 24hr

DIDS USING THIS IN-GROUP:
123456789 mexico df

Campaig

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Hopper Level: 1
Allow Inbound and Blended: Y
Allow Closers: Y
Active: Y
Auto Dial Level: 1
Allowed Inbound Groups : SALES


[williamconley]

1) This is not a "general discussion" post, it is a "Support" post. I will move it to Support for you, please post in Support next time to get more appropriate readers and responses.

2) use sip debug to find the context that the call was sent to. This should help you determine the problem. Usually the problem is that the call is authenticating to the wrong sip account (or NO sip account) and is then sent to the unauthenticated sip context which is "trunkinbound" and works anyway. However: You are using GoAutoDial which may still be set to have unauthenticated sip inbound calls sent to "default" instead of "trunkinbound", and then the calls fail because there is no matching pattern.

[JCSANHUEZA]

my sip debug...

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<------------->
[Jul  9 10:44:00] NOTICE[2940]: chan_sip.c:7487 sip_reregister:    -- Re-registration for  8780044026@sip.voztele.com.mx
[Jul  9 10:44:00] REGISTER 13 headers, 0 lines
[Jul  9 10:44:00] Reliably Transmitting (NAT) to 217.18.238.38:5062:
REGISTER sip:sip.voztele.com.mx:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.188:5060;branch=z9hG4bK1fa617f4;rport
From: <sip:8780044026@sip.voztele.com.mx>;tag=as0a4ede08
To: <sip:8780044026@sip.voztele.com.mx>
Call-ID: 0a44f9ab1b4050820a6e652c176beb84@127.0.0.1
CSeq: 153 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="8780044026", realm="sip.voztele.com.mx", algorithm=MD5, uri="sip:sip.voztele.com.mx:5062", nonce="4ffafc9aa6ce1363f9095f4b9be0ea725adc0e07", response="3a8eea95224bdae26efc618beefb9ff5"
Expires: 120
Contact: <sip:8780044026@192.168.1.188>
Event: registration
Content-Length: 0


---
[Jul  9 10:44:00]
<--- SIP read from 217.18.238.38:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.188:5060;branch=z9hG4bK1fa617f4;rport=5060
From: <sip:8780044026@sip.voztele.com.mx>;tag=as0a4ede08
To: <sip:8780044026@sip.voztele.com.mx>;tag=ba3d51acad53eeb51d56ab2459dbff7b.9a27
Call-ID: 0a44f9ab1b4050820a6e652c176beb84@127.0.0.1
CSeq: 153 REGISTER
NatPing-OPTIONS: yes
NatPing-Interval: 30
Contact: <sip:8780044026@192.168.1.188:5060>;expires=120
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0


<------------->
[Jul  9 10:44:00] --- (11 headers 0 lines) ---
[Jul  9 10:44:00] Scheduling destruction of SIP dialog '0a44f9ab1b4050820a6e652c176beb84@127.0.0.1' in 32000 ms (Method: REGISTER)
[Jul  9 10:44:00] NOTICE[2940]: chan_sip.c:12628 handle_response_register: Outbound Registration: Expiry for sip.voztele.com.mx is 120 sec (Scheduling reregistration in 105 s)
[Jul  9 10:44:01]   == Parsing '/etc/asterisk/manager.conf': [Jul  9 10:44:01] Found
[Jul  9 10:44:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  9 10:44:01]   == Parsing '/etc/asterisk/manager.conf': [Jul  9 10:44:01] Found
[Jul  9 10:44:01]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  9 10:44:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  9 10:44:01]   == Manager 'sendcron' logged off from 127.0.0.1
[Jul  9 10:44:06]
<--- SIP read from 193.22.119.20:5060 --->
INVITE sip:8780044026@192.168.1.188:5060 SIP/2.0
Record-Route: <sip:193.22.119.20;lr=on;ftag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363>
Record-Route: <sip:193.22.119.20;lr=on;ftag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363>
Record-Route: <sip:200.76.112.13:5070;lr=on;ftag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363>
From: <sip:5551281116@200.76.112.13:5060;user=phone>;tag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363
To: <sip:1401255512090746@200.76.112.13:5070;user=phone>
Call-ID: 6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13
CSeq: 1 INVITE
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK989c.426644e3.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK989c.326644e3.0
Via: SIP/2.0/UDP 200.76.112.13:5070;branch=z9hG4bK989c.25d3bf53.0
Via: SIP/2.0/UDP 200.76.114.200:5060;branch=z9hG4bK-82e363-ff483c03-37f08a6f
Max-Forwards: 28
Contact: <sip:5551281116@200.76.114.200:5060;user=phone>
Min-SE: 90
Content-Type: application/SDP
Content-Length: 327
X-CPLFROM: 8780044026

v=0
o=200.76.114.200 1341852543 1341852543 IN IP4 200.76.114.200
s=Session SDP
c=IN IP4 200.76.114.212
t=0 0
m=audio 51780 RTP/AVP 18 8 0 15 17 96
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:15 G728/8000
a=rtpmap:17 T38/8000
a=fmtp:96 0-15
a=rtpmap:96 telephone-event/8000
a=ptime:20

<------------->
[Jul  9 10:44:06] --- (18 headers 14 lines) ---
[Jul  9 10:44:06] Sending to 193.22.119.20 : 5060 (NAT)
[Jul  9 10:44:06] Using INVITE request as basis request - 6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13
[Jul  9 10:44:06] Found no matching peer or user for '193.22.119.20:5060'
[Jul  9 10:44:06] Found RTP audio format 18
[Jul  9 10:44:06] Found RTP audio format 8
[Jul  9 10:44:06] Found RTP audio format 0
[Jul  9 10:44:06] Found RTP audio format 15
[Jul  9 10:44:06] Found RTP audio format 17
[Jul  9 10:44:06] Found RTP audio format 96
[Jul  9 10:44:06] Peer audio RTP is at port 200.76.114.212:51780
[Jul  9 10:44:06] Found audio description format G729 for ID 18
[Jul  9 10:44:06] Found audio description format PCMA for ID 8
[Jul  9 10:44:06] Found audio description format PCMU for ID 0
[Jul  9 10:44:06] Found unknown media description format G728 for ID 15
[Jul  9 10:44:06] Found unknown media description format T38 for ID 17
[Jul  9 10:44:06] Found audio description format telephone-event for ID 96
[Jul  9 10:44:06] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Jul  9 10:44:06] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Jul  9 10:44:06] Peer audio RTP is at port 200.76.114.212:51780
[Jul  9 10:44:06] Looking for 8780044026 in default (domain 192.168.1.188)
[Jul  9 10:44:06]
<--- Reliably Transmitting (NAT) to 193.22.119.20:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK989c.426644e3.0;received=193.22.119.20
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK989c.326644e3.0
Via: SIP/2.0/UDP 200.76.112.13:5070;branch=z9hG4bK989c.25d3bf53.0
Via: SIP/2.0/UDP 200.76.114.200:5060;branch=z9hG4bK-82e363-ff483c03-37f08a6f
From: <sip:5551281116@200.76.112.13:5060;user=phone>;tag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363
To: <sip:1401255512090746@200.76.112.13:5070;user=phone>;tag=as0cde4cd2
Call-ID: 6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Jul  9 10:44:06] NOTICE[2940]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '8780044026' rejected because extension not found.
[Jul  9 10:44:06] Scheduling destruction of SIP dialog '6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13' in 32000 ms (Method: INVITE)
[Jul  9 10:44:06]
<--- SIP read from 193.22.119.20:5060 --->
ACK sip:8780044026@192.168.1.188:5060 SIP/2.0
Via: SIP/2.0/UDP 193.22.119.20;branch=z9hG4bK989c.426644e3.0
From: <sip:5551281116@200.76.112.13:5060;user=phone>;tag=1ff7ae0-c8724cc8-13c4-82e363-12ca4bbe-82e363
Call-ID: 6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13
To: <sip:1401255512090746@200.76.112.13:5070;user=phone>;tag=as0cde4cd2
CSeq: 1 ACK
User-Agent: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0


<------------->
[Jul  9 10:44:06] --- (8 headers 0 lines) ---
[Jul  9 10:44:06] Really destroying SIP dialog '6c07a48-c8724cc8-13c4-82e363-58b7e1c1-82e363@200.76.112.13' Method: ACK
[Jul  9 10:44:06]   == Parsing '/etc/asterisk/manager.conf': [Jul  9 10:44:06] Found
[Jul  9 10:44:06]   == Manager 'sendcron' logged on from 127.0.0.1
[Jul  9 10:44:06]   == Manager 'sendcron' logged off from 127.0.0.1

VICIDIAL*CLI>

Thanks Mr. williamconley

[williamconley]

i did not mean "any old sip debug that pops up". you'll have to find the sip debug directly related to the rejected call. it will immediately precede the rejection AND contain pertinent information regarding this particular call and why it is rejected.

[striker]

hi

[Jul 8 19:59:18] NOTICE[2944]: chan_sip.c:14035 handle_request_invite: Call from '' to extension 'XXXXXXXXX'' rejected because extension not found.

is that showing incomming DID as XXXXXXXXX or showing the exact number?

may be your incomming did comes in default context put the below dialplan in context default

[default]

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

NOte: XXXXXXXXX change this to ur exact DID .

[williamconley]

hi

[Jul 8 19:59:18] NOTICE[2944]: chan_sip.c:14035 handle_request_invite: Call from '' to extension 'XXXXXXXXX'' rejected because extension not found.

is that showing incomming DID as XXXXXXXXX or showing the exact number?

may be your incomming did comes in default context put the below dialplan in context default

[default]

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

NOte: XXXXXXXXX change this to ur exact DID .

Also note: This will require a new entry for each DID entered.

It also does not need to be entered in "default" directly in extensions.conf: It can be entered beneath any carrier along with the rest of the dial plan (leave a blank line if you would like to separate it visually). But will only work if the dialplan is actually looking in default. If it's looking in "gafachi-incoming" or something equally interesting, it still won't work until it is created or merely fixed.