[asterisk-users] IVR and DTMF

Thu Apr 9 23:54:02 CDT 2009

REPOSTED with MORE Info and Modified Subject Line:
--------------------------------------------------------

I am using one of the Minute Provider to dial out USA numbers.

Now in one of my process, we need to Dial IVR and the enter DTMF digit and
then it connects to the automated IVR.

When I dial out the IVR directly using Xlite and VOIP Mins provider , it
works perfectly. but when In try from Dialer, after entering 10 digit
customer number, it says TRANSFERRING, THERE IS NOONE IN THE SESSION. .!!!

How to solve it ?

[sip216]
type=peer
username=11XX
fromuser=11XX
authuser=11XX
secret=LakshmiXX
host=216.128.XX.X
fromdomain=216.128.XX.X
nat=no
canreinvite=no
insecure=very
disallow=all
allow=g729
allow=ulaw
context=default
dtmfmode=rfc2833

Its like...

1) We call up our Customer
2) We convince the customer to buy something
3) Customer Agrees.
4) For security reason, we need to record everything on 3rd party server
5) We keep the customer on hold
6) We dial a 727 series number
7) We enter the Access Code and Room Number
[image: Cool] Then we enter the Customer Phone Number
9) Then IVR start with automated script.
10) Completes the Verification through
11) Generates the Unique code
12) Call Ends.

We cant REACH 9th step.

Regarding CLI :

*Quote:*

[root at vicidialnow ~]# asterisk -r
Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for
details.
This is free software, with components licensed under the GNU General
Public
License version 2 and other licenses; you are welcome to redistribute it
under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2615)
Verbosity is at least 21
-- Executing AGI("SIP/cc101-b7a07420", "agi://127.0.0.1:4577/call_log") in
new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7a07420", "SIP/17275691533 at sip8||tTor") in new
stack
-- Called 17275691533 at sip8
-- SIP/sip8-0825f9b0 is making progress passing it to SIP/cc101-b7a07420
-- SIP/sip8-0825f9b0 answered SIP/cc101-b7a07420
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 817275691533, 2) exited non-zero on
'SIP/cc101-b7a07420'
-- Executing DeadAGI("SIP/cc101-b7a07420", "agi://127.0.0.1:4577/call_log")
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("SIP/cc101-b7a07420", "agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45)")
in new stack
-- AGI Script
agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45)
completed, returning 0
vicidialnow*CLI>

sip debug shows below lines:

*Quote:*

--- (12 headers 0 lines) ---
Sending to 192.168.0.50 : 12714 (NAT)
Transmitting (NAT) to 192.168.0.50:12714:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:12714
;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714
From: "cc106"<sip:cc106 at 192.168.0.2 <sip%3Acc106 at 192.168.0.2>>;tag=7f1cff22
To: "817275691533"<sip:817275691533 at 192.168.0.2<sip%3A817275691533 at 192.168.0.2>
>;tag=as02559696
Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:817275691533 at 192.168.0.2 <sip%3A817275691533 at 192.168.0.2>>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

---
Scheduling destruction of call
'617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx' in 32000 ms
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060:
BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport
From: "cc106" <sip:fiddialer at 59.xxx.xx.xx>;tag=as3f9466a7
To: <sip:17275691533 at 8.14.xxx.xxx>;tag=1902000923108720995156225
Call-ID: 617ad67d47db8e4a2155fcd51d1089ff at 59.xxx.xx.xx
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
== Spawn extension (default, 817275691533, 2) exited non-zero on
'SIP/cc106-b7a1a9d0'
-- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://127.0.0.1:4577/call_log")
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)")
in new stack
-- AGI Script
agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12)
completed, returning 0
Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.'
vicidialnow*CLI>
<-- SIP read from 8.14.xxx.xxx:5060:
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport
From: "V0219160007000134649" <sip:fiddialer at 59.xxx.xx.xx>;tag=as79fae976
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:8.14.xxx.xxx:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 225

v=0
o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx
s=VoipSIP
i=Audio Session
c=IN IP4 8.14.xxx.xxx
t=0 0
m=audio 6220 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 11 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 8.14.xxx.xxx:6220
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0
(nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp>
set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for
address/port to send to
set_destination: set destination to 8.14.xxx.xxx, port 5060
Transmitting (no NAT) to 8.14.xxx.xxx:5060:
ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport
From: "V0219160007000134649" <sip:fiddialer at 59.xxx.xx.xx>;tag=as79fae976
To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221
Contact: <sip:fiddialer at 59.xxx.xx.xx>
Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Let me know if any other information is required
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090410/9e243138/attachment.htm 