How to setup IVR

General Help
1

Hi

I am new to asterisk, I need to setup ivr and route all inbound calls to IVR. Could you please help me to to that.

thanks in advance

2

Please see this for first step:
http://wiki.freepbx.org/display/FPG/IVR+Module+User+Guide

3

Thanks, let me check

4

I’ve creaed ivr now,how to route all the inbound calls to ivr?

5

And for step two:
http://wiki.freepbx.org/display/FPG/Inbound+Route+User+Guide

6

Do you have any samples for the above?

7

please follow above guide, you should spend more time!

PS: if you don’t know what is going on, you can’t solve any issue that may happen

8

Hi,

I am using GrandStream HT 503 ATA. I am not getting caller from HT503.

Please help me to resolve this issue?

My setup is:

Analog PBX =>HT503=>Asterisk

9

Gonna have to post more information for people to help you out in any way

10

Hi jcroy,

FXO PORT.png1366×3783 818 KB
Basic Settings.png1366×1849 176 KB

Trunk context:

type=friend
fromuser=2008
secret=asdf1234
context=from-trunk
dtmfmode=rfc2833
host=dynamic
nat=no
trustrpid=yes
sendrpid=yes
canreinvite=no
insecure=yes
qualify=yes
disallow=all
allow=ulaw&alaw

11

The above is my HT 503 FXO and Basic Settings screenshot. and i have shared my trunk context also. Please check

12

Assuming you have a default FreePBX installation, 7777 is the number that you need to forward your calls to from the FXO port.

In basic settings, replace 4000 with 7777 on “unconditional call forward to voip” and you should start receiving calls, assuming you have also configured your HT503 for one-stage dialling.

13

Hi,

It is not a default freepbx installation. I have created inbound route and Trunk. My inbound route no is 4000.
Previously it was working fine, suddenly it stops sending caller id to freepbx. Some times it is sending and some time it is not.

Below is my Grandstream debug log

Ôò¡ Ü ½S•Y9l J J ‚‡“¸’ë¿j E`<°B @OÁ
ÄÄ(z‡OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 30.30.30.10:5060;branch=z9hG4bK6a40bf43
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as74a90f28
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(11.21.0)
Date: Thu, 17 Aug 2017 08:28:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

¾S•YyL D D ‚‡“¸’ë¿j E`6°Ž @O{
ÄÆ"zOPTIONS sip:[email protected]:5062 SIP/2.0
Via: SIP/2.0/UDP 30.30.30.10:5060;branch=z9hG4bK7a5f2302
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as6f4376ce
To: sip:[email protected]:5062
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.190.7(11.21.0)
Date: Thu, 17 Aug 2017 08:28:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

ÊS•Y‘ ) ) ‚‡“¸’ë¿j E`e±H @NÜ
ÄÆazfSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 30.30.30.8:5062;branch=z9hG4bK757094338;received=30.30.30.8;rport=5062
From: sip:[email protected];tag=428012857
To: sip:[email protected]:5060;tag=as7af3b7fb
Call-ID: [email protected]
CSeq: 30 INVITE
Server: FPBX-13.0.190.7(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3fa10c1a"
Content-Length: 0