Configuring Inbound Trunk

[nikeshshk]

Hi Vicidial Experts

I need to add new DID inbound trunk. I went through Managers Manual but still inbound is not working in my system.

I did created new In-Group, new did entry, new inbound campign.

do I have to make changes in sip.conf and extenstions.conf.


DID details from my DID provider
DID number:123456789
DID username:555555
DID password: pppppp
DID host: 127.0.0.1

Your Earliest response will be highly appreciated.

Thanks
Nikesh

[williamconley]

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

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.

Similar to This:
Vicibox X.X from .iso | Vicidial X.X.X Build XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

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)
____________

Post your account entry for this carrier. Did you add "context=trunkinbound" to the account entry?

Have you successfully registered to your SIP provider?

The goal is to get some sort of activity as a result of an inbound call in your system. Even if the only activity is your system rejecting the call, that's at least evidence that the call is being sent to your system.

Post your sip debug for the moment a call fails to come in. Do not post 3000 lines of code. Try to do this when the system is NOT in use. Just post whatever activity results from the "failed" call.

If there is NO activity directly related to the failed inbound call, you have a problem: if the call never gets to your server, your server will never be able to handle the call. So first your goal is to get SOME activity in the sip debug CLI.

[nikeshshk]

Firstly I apologize for incomplete post

This time I have provided all details here.

vicidial version:goautodial-ce-2.0-rc2

asterisk version:Asterisk 1.4.27.1-1
telephony hardware: Intel Core2 duo, 4GB RAM, 500 GB HDD
Extra Software in server: iptraf
Installation Method: General installation as described http://carlo.taguinod.net/vicidialnow/v ... -guide.pdf
Single Server | No Digium/Sangoma Hardware


My Sip.conf configuration
[inbound]
disallow=all
allow=ulaw
username=555555
secret=pppppp
type=friend
host=127.0.0.1
fromdomain=127.0.0.1
fromuser=555555
qualify=1000
context=trunkinbound
dtmfmode=rfc2833
permit=0.0.0.0/0.0.0.0

register => 555555:pppppp@127.0.0.1/61212347456


My extensions.conf

[default]
exten => 212347456,1,Ringing
exten => 212347456,2,Wait(1)
exten => 212347456,3,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----Support-----1111-----Closer-----park----------999-----1)
exten => 212347456,4,Hangup



sip show peers
Name/username Host Dyn Nat ACL Port Status
inbound/555555 127.0.0.1 N A 5060 OK (330 ms)




<--- SIP read from 67.212.164.231:5060 --->
SIP/2.0 200 OK
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 67.212.164.231:5060;branch=z9hG4bK1d441498;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as14a3043d
Call-ID: 583214a00b95b1ea0563f00c0e2a7dda@192.168.0.5
To: <sip:67.212.164.231;cpd=on>;tag=011141100013
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '583214a00b95b1ea0563f00c0e2a7dda@192.168.0.5' Meth od: OPTIONS
Reliably Transmitting (NAT) to 192.168.0.212:8266:
OPTIONS sip:cc101@192.168.0.212:8266;rinstance=11e646707b96d847;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK76124f63;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as41718175
To: <sip:cc101@192.168.0.212:8266;rinstance=11e646707b96d847;cpd=on>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 66fd157f24ac748e0f5a00c71e855db0@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 29 Oct 2010 15:13:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
go*CLI>
<--- SIP read from 192.168.0.212:8266 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK76124f63;rport=5060
Contact: <sip:192.168.0.212:8266>
To: <sip:cc101@192.168.0.212:8266;rinstance=11e646707b96d847;cpd=on>;tag=fb66662 b
From: "asterisk"<sip:asterisk@192.168.0.5>;tag=as41718175
Call-ID: 66fd157f24ac748e0f5a00c71e855db0@192.168.0.5
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O
ser-Agent: eyeBeam release 1013t stamp 43070
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---

<--- SIP read from 192.168.0.212:8266 --->



<------------->
Really destroying SIP dialog '66fd157f24ac748e0f5a00c71e855db0@192.168.0.5' Meth od: OPTIONS

Thanks a lot in advance

[williamconley]

username=555555
secret=pppppp
type=friend
host=127.0.0.1
fromdomain=127.0.0.1
fromuser=555555

i am assuming that you username is not 55555, but that you masked it to avoid posting confidential information. (ordinarily i recommend just changing secret stuff to XXXX)

But I am not assuming that you changed the host to 127.0.0.1 from the real ip address. If you have a host of 127.0.0.1, the system has no idea where the call is coming from to match it to this carrier to send it to trunkinbound ... which will never work. (if you DID change it, please next time change it to something "obviously masked" like XXXX )

On the other hand, i don't see anything in your sip debug that is related to an inbound call. I see something from cc101, but that's a vicidial agent, not an inbound did. As I said, until you get SOME result on the asterisk sip debug cli from dialing the inbound number, you cannot make vicidial manage the call (it's not here, it cannot be managed)

Also you didn't answer:

Have you successfully registered to your SIP provider?

Which is Very handy information. So keep tryin'!

[nikeshshk]

Sorry for confusion again william

username=xxxx
secret=xxxx
type=friend
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
fromuser=xxxxxx


registration to sip server is sucessfull here is my sip show peers result

sip show peers
Name/username Host Dyn Nat ACL Port Status
inbound/xxxxx xxx.xxx.xxx.xxx N A 5060 OK (330 ms)

I assume Call is not getting landed in my vicidial box. is my configuration file sip.conf and extension.conf in correct order?

or is there some other configuration file that need to be updated too?

Thanks

[nikeshshk]

But I am not assuming that you changed the host to 127.0.0.1 from the real ip address. If you have a host of 127.0.0.1, the system has no idea where the call is coming from to match it to this carrier to send it to trunkinbound ... which will never work. (if you DID change it, please next time change it to something "obviously masked" like XXXX )

I did replaced 127.0.0.1 with my sip servers ip address

[robin]

Is the number you use for you DID correct?

If I look at my own system, the inbound dids have to be without a leading 0.
Did you check what happens in CLI when you dial in? That might give you some more info about where the incoming call is routed to.

[williamconley]

if you have registered successfully, you should get activity related to an inbound call in your sip debug output. if you do not, you must contact your sip provider to find out why or give us more information related to how you route calls on your sip provider interface.

until the call GETS to your asterisk server, your asterisk server cannot manage the call.

the first goal always has to be to get the call to asterisk. after that it will work, without that it will never work. (the cab driver cannot take you to the airport if you are not in the cab )

[covarrubiasgg]

A common issue that i got when i am settnig up a new carrier, is that even when i set context=trunkinbound the call keeps coming to default, so i can see on my CLI,

¨NOTICE: Call rejected from xxx.xxx.xxx.xxx, extension not found in context¨


As robin sugested you, Open your CLI, call your DID and check for any Notice, Warning or Error.

[williamconley]

this can occur based on the path the call takes to get to you: it must authenticate to a sip context that will send it to trunkinbound, if it does not ... asterisk will send it to the context pointed to by the sip account it landed in. this can be based on user/pass or ip address, but is usually ip address for inbound calls.