Incomig Calls (Inboud Route) To IVR

General Help
1

i need to configure inbound Route To IVR

am try to do that but always that go to voicemail

am have latest version of FreePBX that released in 4-may-2012

hope to help me please

2

if you have all your other and outbound routes and your trunks set up and you can make and receive calls correctly then there is nothing extra that you would have to add to the trunk settings.

Having the voicemail enabled or disabled shouldn’t make a difference as far as I know as when i was testing my own system I had 90+ extensions all with voicemail enabled and once the inbound route was set up it worked fine.

If its only the IVR that cannot be reached then it would still appear to me that the inbound route to the IVR is not set up correctly

3

Assuming that you have already created the IVR that you want to use, this is the way I set the IVR and it works.

  1. Create an extension and call it IVR, reception or whatever, make sure you give it an extension number.
  2. Go to INBOUND ROUTES and select add incoming route
  3. Set the DID number to the extension number you created for the IVR
  4. Scroll down the page to SET DESTINATION and select IVR from the drop down menu and then select the recording you want to use from the drop down menu next to that

Hopefully this helps

4

First of all

thanks for your help

but when am try to call that go to voicemail System

first of all there is way to

  • is their data that should be filled inside Incoming SIP Trunk

  • disable all voicemail to check that

5

Send log when calling the DID that is mapped to the IVR. The digits your carrier is sending must not match what you have in the route.

6

but hope to check this with me

  • Go to system Recording and make recording for IVR name it (IVR_Recording)

  • make IVR (IVR1) and Select IVR with (IVR_Recording)

  • Make extension number 199

  • go to inbound route --> Add incoming route

  • in DID fill 199

this all Setting but still when am ringing listen Voicemail message

–which log file that need to check if that work correctly or not –

7

You need to turn log level up ‘asterisk -r core set verbose 128’

Check log file /var/log/asterisk/full

8

thanks for you help

but when am check this is voicmail from provider

when am connect DID to X-lite Directly its work

but that not work inside FreePBX

so now , outbound work , inbound not work

9

[2012-05-17 06:12:12] WARNING[2819]: chan_sip.c:14795 check_auth: username mismatch, have , digest has <>
[2012-05-17 06:12:12] NOTICE[2819]: chan_sip.c:23238 handle_request_invite: Failed to authenticate device “test” sip:[email protected]:5060;tag=1012173178074442

10

the calls delivered but with failed logs

the logs are when no 063001333 try to call my number 63000882

[2012-05-17 06:19:12] WARNING[2819]: chan_sip.c:14795 check_auth: username mismatch, have , digest has <>
[2012-05-17 06:19:12] NOTICE[2819]: chan_sip.c:23238 handle_request_invite: Failed to authenticate device “test” sip:[email protected]:5060;tag=1019183178495208

hope to help please

11

It seems that you dont have your trunk set correctly. Your provider is expecting an authuser and it looks as if you are not sending one. Make sure you have authuser=XXXXXXX set in your peer details and the authuser included in your register string like this username:secret:authuser@host/username.

12

Same issue

am add authuser but same problem

My Sip trunk Setting as following

Outgoing Settings

host=tarasol.jo
username=63000882
authuser=63000882
secret=XXXXXX
type=peer

#############################################

Incoming Settings

context=from-trunk
dtmfmode=rfc2833
host=tarasol.jo
secret=XXXXXX
type=user
username=63000882
authuser=63000882
###########################################

Register String

63000882:XXXXXX:[email protected]/63000882

13

after add

insecure=invite

for incoming calls

thats work fine

14