vicidial.org

I have an inbound carrier setup with account entry:

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[didlogic-trunk]
host=177.77.100.11
username=username
secret=passwrd
type=friend
insecure=port,invite
context=888inbound

I created in Inbound:
DID, DID Extension: 13605551212,DID Route: IN_GROUP, User Route Settings In-Group: 888 Inound, In-Group ID: 888 Inbound
In-Groups, Group ID: 13605551212, Group name: 888 Inbound, Default Tranfer Group: 888 Inbound
In Campaign it is set to Allow inbound and blended: Y, Allowed Inbound Groups: 888 Inbound checked

in extensions.conf I have:

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[888inbound]
;agent dial in:
;exten => _13605551212,1,Answer  ;Answer the line
;exten => _13605551212,2,AGI(agi-AGENT_dial_in.agi)
;exten => _13605551212,3,Hangup
 
; DID call routing process
exten => _13605551212,1,AGI(agi-DID_route.agi)  ; use this one instead of the one below if you are having
;delay issues, and match to number of received digits
;exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
;exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


and here's what I get from asterisk:

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[May 31 20:10:34]
<--- SIP read from 177.77.100.11:5060 --->
INVITE sip:13605551212@111.1.111.11 SIP/2.0
Via: SIP/2.0/UDP 177.77.100.11:5060;branch=z9hG4bK049da6da;rport
Max-Forwards: 70
From: "KRIS ALLEN" <sip:17275551212@177.77.100.11>;tag=as460a699a
To: <sip:13605551212@111.1.111.11>
Contact: <sip:17275551212@177.77.100.11>
Call-ID: 2b459d7f49b61de66de3b82c5920715b@177.77.100.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4
Date: Fri, 01 Jun 2012 00:10:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 320

v=0
o=root 1840196988 1840196988 IN IP4 177.77.100.11
s=Asterisk PBX 1.6.2.9-2+squeeze4
c=IN IP4 177.77.100.11
t=0 0
m=audio 16432 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------->
[May 31 20:10:34] --- (14 headers 14 lines) ---
[May 31 20:10:34] Sending to 177.77.100.11 : 5060 (NAT)
[May 31 20:10:34] Using INVITE request as basis request - 2b459d7f49b61de66de3b82c5920715b@177.77.100.11
[May 31 20:10:34] Found peer 'VerInbound'
[May 31 20:10:34] Found RTP audio format 0
[May 31 20:10:34] Found RTP audio format 8
[May 31 20:10:34] Found RTP audio format 18
[May 31 20:10:34] Found RTP audio format 101
[May 31 20:10:34] Found audio description format PCMU for ID 0
[May 31 20:10:34] Found audio description format PCMA for ID 8
[May 31 20:10:34] Found audio description format G729 for ID 18
[May 31 20:10:34] Found audio description format telephone-event for ID 101
[May 31 20:10:34] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[May 31 20:10:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 31 20:10:34] Peer audio RTP is at port 177.77.100.11:16432
[May 31 20:10:34] Looking for 13605551212 in vinbound (domain 111.1.111.11)
[May 31 20:10:34]
<--- Reliably Transmitting (NAT) to 177.77.100.11:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 177.77.100.11:5060;branch=z9hG4bK049da6da;received=177.77.100.11;rport=5060
From: "KRIS ALLEN" <sip:17275551212@177.77.100.11>;tag=as460a699a
To: <sip:13605551212@111.1.111.11>;tag=as0b2a28de
Call-ID: 2b459d7f49b61de66de3b82c5920715b@177.77.100.11
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[May 31 20:10:34] NOTICE[3027]: chan_sip.c:15566 handle_request_invite: Call from '43282' to extension '13605551212' rejected because extension not found.
[May 31 20:10:34] Scheduling destruction of SIP dialog '2b459d7f49b61de66de3b82c5920715b@177.77.100.11' in 8512 ms (Method: INVITE)


What am I missing?

the part I see that wierds me out is in asterisk:

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[May 31 20:10:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May 31 20:10:34] Peer audio RTP is at port 177.77.100.11:16432
[May 31 20:10:34] Looking for 13605551212 in vinbound (domain 111.1.111.11)

I have another inbound line using vinbound in it's context= how would this inbound DID be crossing with that one? The carrier description shows its right.
for info on vinbound, I have Carrier ID VerInbound with account entry:

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[VerInbound]
host=177.77.100.11
username=username
secret=passwd
type=friend
insecure=port,invite
context=vinbound

comes from the same host because I have several DIDs purchased through this company but other than that this inbound line works the other doesn't....

It doesn't really matter. Change them all to context=trunkinbound as that is the only path to allow inbound calls to be controlled by Vicidial. Once they are all sent to trunkinbound you can control them and it really makes no difference which carrier they came from.

I have seen this with clients who modify their sip.conf file and it often turns out to be an interesting but odd result of an unusual combination of settings. You should be using the "stock" sip.conf file and the generated sip-vicidial.conf should have context=trunkinbound for all account entries. Then you can control everything in the GUI and not care about why. If you are determined to work out why, this is an Asterisk question purely and not really a Vicidial question unless the sip-vicidial.conf contains a flaw ... (or the provided sample sip.conf contains a flaw). But since these work for everyone else quite well ...

Revisiting this problem again...
Everything appears setup. Carrier, DID, Ingroup, Campaign...
never the less:

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[Sep 24 19:19:20]     -- Executing [3025551212@trunkinbound:1] AGI("SIP/manchesterd-00001279", "agi-DID_route.agi") in new stack
[Sep 24 19:19:20]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Sep 24 19:19:20]     -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20120924191920_3025551212_7275551212)
[Sep 24 19:19:20]     -- AGI Script agi-DID_route.agi completed, returning 0
[Sep 24 19:19:20]     -- Executing [99909*6***DID@default:1] Answer("SIP/manchesterd-00001279", "") in new stack
[Sep 24 19:19:20]     -- Executing [99909*6***DID@default:2] AGI("SIP/manchesterd-00001279", "agi-VDAD_ALL_inbound.agi") in new stack
[Sep 24 19:19:20]     -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Sep 24 19:19:21]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Sep 24 19:19:21]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Sep 24 19:19:21]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Sep 24 19:19:21]     -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Sep 24 19:19:22]     -- AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Sep 24 19:19:22]     -- Sent into invalid extension '85026666666666' in context 'default' on SIP/manchesterd-00001279
[Sep 24 19:19:22]     -- Executing [i@default:1] Playback("SIP/manchesterd-00001279", "invalid") in new stack
[Sep 24 19:19:22]     -- <SIP/manchesterd-00001279> Playing 'invalid' (language 'en')

Carrier Settings:
Account Entry:

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[manchesterd]
type=friend
host=*myhost*
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
qualify=yes
username=3025551212
context=trunkinbound


Tried several dialplans:

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exten => _3025551212,1,NoOp(Set(CALLERID(number)=3023949707))
exten => _3025551212,n,AGI(agi://127.0.0.1:4577/call_log)
exten => _3025551212,n,Dial(${DIAL6TRUNK}/${EXTEN:1},,Ttor)
exten => _3025551212,n,Hangup

to

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exten => _3025551212,1,Answer  ;Answer the line
exten => _3025551212,2,AGI(agi-DID_route.agi)
exten => _3025551212,3,Hangup

Nothing but "I am sorry that is not a valid extension"

DID points to In_Group, User Route Settings In-Groupe:AgentInbound
AgentInbound points to Default Transfer Group AgentInbound
Campaign settings allow inbound, and inbound is selected... what am I missing?

"Sent into invalid extension '85026666666666' in context 'default' on SIP/manchesterd-00001279"

check your ingroup settings , somewhere it is dialling 85026666666666.