GOautodial Open Source Omni-channel Contact Center Suite

1. make sure you have proper carrier to receive a call from a DID
2. Create a phone with Voicemail enabled.foreg: phone no 100
3. upload your voice file to be played for eg: welcome.wav in server /var/lib/asterisk/sounds or use goautodial audio store
4. Now create a New DID in goautodial and Point the Did route to extensions and putt 999 in extension field
5. Enable the Custom dialplan in server settings and enter the below dial plan

exten => 999,1,Answer
exten => 999,2,Playback(welcome)
exten => 999,3,Voicemail(100)
exten => 999,4,Hangup

reload and check

br
striker
www.striker24x7.blogspot.com

Hi Striker,

Thanks for the reply that does make sense!

What I'm struggling with to start ( Just to test the incoming number ) is directing it straight to an extension / phone. (Which will need for admin numbers regardless)

Belows the report from sip set debug on for the inbound call (Which just gives me a ladies voice saying not available)

<------------->
[Jul 9 11:17:26] --- (6 headers 0 lines) ---
[Jul 9 11:17:26]
<--- SIP read from UDP:203...69:5060 --->
INVITE sip:07XXXXXXXX@123..66:5060 SIP/2.0
Via: SIP/2.0/UDP 203..69:5060;branch=z9hG4bKelsa0a0010r13fp40061.1
From: <sip:04XXXXXXXX@voice.mibroadband.com.au;user=phone>;tag=SDpcuhe01-975378952-1404868646587-
To: "Richard Hubbard"<sip:07XXXXXXXX@voice.mibroadband.com.au>
Call-ID: SDpcuhe01-80e57d08c03384cc1d7f93e367cbb70b-au418e3
CSeq: 207170398 INVITE
Contact: <sip:SD1o6i6-vv9pmjj9mvp7tbr5iqonkdpvku9rouvrjrdrvorsmqvtoh8gjpv1-6@203..69:5060;transport=udp>
P-Called-Party-ID: <sip:07XXXXXXXX@voice.mibroadband.com.au>
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept: application/media_control+xml,application/sdp,multipart/mixed
Supported: timer
Min-SE: 60
Session-Expires: 1800;refresher=uas
Max-Forwards: 9
Content-Type: application/sdp
Content-Length: 376

v=0
o=BroadWorks 35983302 1 IN IP4 203..69
s=-
c=IN IP4 203..69
t=0 0
m=audio 18022 RTP/AVP 8 18 0 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:4 bitrate=6.3
a=fmtp:101 0-15
a=silenceSupp:off - - -
a=sendrecv
a=bsoft: 1 image udptl t38
<------------>
[Jul 9 11:17:26] --- (16 headers 17 lines) ---
[Jul 9 11:17:26] Sending to 203.161.164.69:5060 (NAT)
[Jul 9 11:17:26] Using INVITE request as basis request - SDpcuhe01-80e57d08c03384cc1d7f93e367cbb70b-au418e3
[Jul 9 11:17:26] No matching peer for '04XXXXXXXX' from '203.161.164.69:5060'
[Jul 9 11:17:26]
<--- Reliably Transmitting (NAT) to 203..69:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 203..69:5060;branch=z9hG4bKelsa0a0010r13fp40061.1;received=203..69;rport=5060
From: <sip:04XXXXXXXXX@voice.mibroadband.com.au;user=phone>;tag=SDpcuhe01-975378952-1404868646587-
To: "Richard Hubbard"<sip:07XXXXXXXX@voice.mibroadband.com.au>;tag=as4cde3aa9
Call-ID: SDpcuhe01-80e57d08c03384cc1d7f93e367cbb70b-au418e3
CSeq: 207170398 INVITE
Server: Asterisk PBX 1.8.23.0-1_centos5.go RPM by demian@goautodial.com
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="49d4f3df"
Content-Length: 0

<------------>

Obviously remvoed some IP's and inbound/outbound numbers.

But what would I put for the SIP Extension, 07XXXXXXXX ? I've tried this and hasn't worked or routed anywhere.

Tried routing to agent, custom extension, phone and all has the same results.

It looks so simple, so what could I possibly be missing?

post cli log without debug

sip set debug off

then make a call and post the cli

br
striker
www.striker24x7.blogspot.com