Inbound Issue (extension)

[udfxrookie]

Having inbound issues... I created a carrier as follows:

Code: Select all

[didlogic-trunk]
host=sip.didlogic.com
username=12345
secret=password
type=friend
insecure=port,invite
context=inbound

Nothing in dialplan entry, but set to active.
Then I create an In group called Inbound, create a DID with extension 18883110274 for the main name and in DID Route send it to the ingroup.
In the campaign it is set to accept inbound calls and I have the ingroup selected. I have one agent with the ingroup selected but when I make a call here are the results:

Code: Select all

[May  6 15:59:34] NOTICE[3032]: chan_sip.c:15566 handle_request_invite: Call from '12345' to extension '18883110274' rejected because extension not found.


Here's with debug on:

Code: Select all

<------------->
[May  6 15:50:51]
<--- SIP read from 178.63.100.24:5060 --->
INVITE sip:18883110274@100.1.0.100 SIP/2.0
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK2402dddb;rport
Max-Forwards: 70
From: "17275551212" <sip:17275551212@178.63.100.24>;tag=as7011d850
To: <sip:18883110274@100.1.0.100>
Contact: <sip:17275551212@178.63.100.24>
Call-ID: 7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4
Date: Sun, 06 May 2012 19:50:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 709352616 709352616 IN IP4 178.63.100.24
s=Asterisk PBX 1.6.2.9-2+squeeze4
c=IN IP4 178.63.100.24
t=0 0
m=audio 12228 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  6 15:50:51] --- (14 headers 14 lines) ---
[May  6 15:50:51] Sending to 178.63.100.24 : 5060 (NAT)
[May  6 15:50:51] Using INVITE request as basis request - 7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24
[May  6 15:50:51] Found peer 'didlogic-trunk'
[May  6 15:50:51] Found RTP audio format 0
[May  6 15:50:51] Found RTP audio format 8
[May  6 15:50:51] Found RTP audio format 18
[May  6 15:50:51] Found RTP audio format 101
[May  6 15:50:51] Found audio description format PCMU for ID 0
[May  6 15:50:51] Found audio description format PCMA for ID 8
[May  6 15:50:51] Found audio description format G729 for ID 18
[May  6 15:50:51] Found audio description format telephone-event for ID 101
[May  6 15:50:51] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
[May  6 15:50:51] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[May  6 15:50:51] Peer audio RTP is at port 178.63.100.24:12228
[May  6 15:50:51] Looking for 18883110274 in inbound (domain 100.1.0.100)
[May  6 15:50:51]
<--- Reliably Transmitting (NAT) to 178.63.100.24:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK2402dddb;received=178.63.100.24;rport=5060
From: "17275551212" <sip:17275551212@178.63.100.24>;tag=as7011d850
To: <sip:18883110274@100.1.0.100>;tag=as70eb4899
Call-ID: 7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[May  6 15:50:51] NOTICE[3032]: chan_sip.c:15566 handle_request_invite: Call from '12345' to extension '18883110274' rejected because extension not found.
[May  6 15:50:51] Scheduling destruction of SIP dialog '7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24' in 8320 ms (Method: INVITE)
[May  6 15:50:51]
<--- SIP read from 178.63.100.24:5060 --->
ACK sip:18883110274@100.1.0.100 SIP/2.0
Via: SIP/2.0/UDP 178.63.100.24:5060;branch=z9hG4bK2402dddb;rport
Max-Forwards: 70
From: "17275551212" <sip:17275551212@178.63.100.24>;tag=as7011d850
To: <sip:18883110274@100.1.0.100>;tag=as70eb4899
Contact: <sip:17275551212@178.63.100.24>
Call-ID: 7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.9-2+squeeze4
Content-Length: 0


<------------->
[May  6 15:50:51] --- (10 headers 0 lines) ---
[May  6 15:50:52] Really destroying SIP dialog '7d761bf608374c9f6a8b9f2845b4613e@178.63.100.24' Method: ACK
[May  6 15:50:57]



Any ideas on what to edit and what to put?

[williamconley]

1) Read The Manual. It's available on EFLO.net. It exists to avoid giving step by step instructions to every user with the same content. Seriously.

2) Welcome to the party! LOL

3) In account settings: context=trunkinbound NOT context=inbound and neither of these settings is related to anything you see or set in the Vicidial GUI. If you set the context to trunkinbound, the call will pass to the Vicidial script managing the trunkinbound context, and the Vicidial GUI settings will then gain control of the call. If you set it to anything else, the call will never enter the control mechanism of Vicidial and this will never work.

4) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

[udfxrookie]

TY, context=trunkinbound
fixed the issue, ty

[williamconley]

Cool. Now go make some money with the FREE Open Source Software (and read the entire manual!)

[wmdbikesh]

Hello having same problem like Call from '' to extension '7114717' rejected because extension not found.


followed managers manual and my configuration are like
host=65.207.151.8
type=peer
disallow=all
allow=g729
insecure=port,invite
context=trunkinbound



and log in asteriskcli is like
[Jun 1 23:44:23] NOTICE[10033]: chan_sip.c:15566 handle_request_invite: Call from '' to extension '7114717' rejected because extension not found.
Scheduling destruction of SIP dialog '75f71c8d68c505765f464582196511a7@63.111.11.139' in 32000 ms (Method: INVITE)

<--- SIP read from 63.111.11.139:5060 --->
ACK sip:7114717@202.70.84.102 SIP/2.0
Via: SIP/2.0/UDP 63.111.11.139:5060;branch=z9hG4bK4a0bf662;rport
From: "asterisk" <sip:asterisk@63.111.11.139>;tag=as5be75642
To: <sip:7114717@202.70.84.102>;tag=as4677f40a
Contact: <sip:asterisk@63.111.11.139>
Call-ID: 75f71c8d68c505765f464582196511a7@63.111.11.139
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '75f71c8d68c505765f464582196511a7@63.111.11.139' Method: ACK
[Jun 1 23:44:26] WARNING[10229]: rtp.c:953 ast_rtcp_read: RTCP Read too short
Really destroying SIP dialog '241fd04b3c0461c4181cecfa17c69f1b@96.31.86.214' Method: OPTIONS
[Jun 1 23:44:29] WARNING[10229]: rtp.c:953 ast_rtcp_read: RTCP Read too short
[Jun 1 23:44:32] WARNING[10229]: rtp.c:953 ast_rtcp_read: RTCP Read too short
[Jun 1 23:44:35] WARNING[10229]: rtp.c:953 ast_rtcp_read: RTCP Read too short



my configuration are
Vicibox 3.10 from ViciBox_Redux.x86_64-3.1.10.iso | Vicidial 2.4-350a Build 111201-0939 | Asterisk 1.4 | Three Servers | No Digium/Sangoma Hardware | No Extra Software After Installation




Please help me

[williamconley]

use sip debug to get the deeper details that precede the message Call from '' to extension '7114717' rejected. That debug information will tell you where the system searched for that extension. Then you will know the path the call took, and you can find out why it went that way instead of where you wanted it to go.

i will note, of course, that the IPs do not match your "host" ip which could indicate that you need to create an "inbound" context for your "inbound" calls from this carrier, so you could place a new host= in the "inbound" context to catch these calls and direct them to trunkinbound as well. this may resolve your likely mismatch.

also: i note you did not post your [xxxxx] context entry from your carrier account entry. that is required, not optional. each carrier must have at least one named context. and they may have several (such as one for in and one for out), but all will contain "context=trunkinbound" to capture calls.