<span class="Apple-style-span" style="font-family: Verdana; font-size: 12px; line-height: 18px; "><div>REPOSTED with MORE Info and Modified Subject Line:</div><div>--------------------------------------------------------</div>
<div><br></div>I am using one of the Minute Provider to dial out USA numbers. <br><br>Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. <br><br>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. .!!! <br>
<br>How to solve it ? <br><br><br>[sip216] <br>type=peer <br>username=11XX <br>fromuser=11XX <br>authuser=11XX <br>secret=LakshmiXX <br>host=216.128.XX.X <br>fromdomain=216.128.XX.X <br>nat=no <br>canreinvite=no <br>insecure=very <br>
disallow=all <br>allow=g729 <br>allow=ulaw <br>context=default <br>dtmfmode=rfc2833</span><br><div><span class="Apple-style-span" style="font-family: Verdana; font-size: 12px; line-height: 18px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; font-size: 12px; line-height: 18px;">Its like... <br>
<br>1) We call up our Customer <br>2) We convince the customer to buy something <br>3) Customer Agrees. <br>4) For security reason, we need to record everything on 3rd party server <br>5) We keep the customer on hold <br>
6) We dial a 727 series number <br>7) We enter the Access Code and Room Number <br><img src="http://www.eflo.net/VICIDIALforum/images/smiles/icon_cool.gif" alt="Cool" border="0"> Then we enter the Customer Phone Number <br>
9) Then IVR start with automated script. <br>10) Completes the Verification through <br>11) Generates the Unique code <br>12) Call Ends. <br><br><br><span style="color: red; ">We cant REACH 9th step.</span><br></span></div>
<div><span class="Apple-style-span" style="color: rgb(255, 0, 0); font-family: Verdana; font-size: 12px; line-height: 18px;"><br></span></div><div><span class="Apple-style-span" style="color: rgb(255, 0, 0); font-family: Verdana; font-size: 12px; line-height: 18px;"><span class="Apple-style-span" style="color: rgb(0, 0, 0); font-size: 16px; line-height: normal; "><span class="postbody" style="font-size: 12px; line-height: 18px; ">Regarding CLI : <br>
<br><br></span><table width="90%" cellspacing="1" cellpadding="3" border="0" align="center"><tbody><tr><td style="font-family: Verdana, Arial, Helvetica, sans-serif; "><span class="genmed" style="font-size: 11px; color: rgb(0, 0, 0); "><b>Quote:</b></span></td>
</tr><tr><td class="quote" style="font-family: Verdana, Arial, Helvetica, sans-serif; font-size: 11px; color: rgb(68, 68, 68); line-height: 125%; background-color: rgb(250, 250, 250); border-top-color: rgb(209, 215, 220); border-right-color: rgb(209, 215, 220); border-bottom-color: rgb(209, 215, 220); border-left-color: rgb(209, 215, 220); border-width: initial; border-style: initial; border-top-style: solid; border-right-style: solid; border-bottom-style: solid; border-left-style: solid; border-left-width: 1px; border-top-width: 1px; border-right-width: 1px; border-bottom-width: 1px; ">
<br><br>[root@vicidialnow ~]# asterisk -r <br>Asterisk 1.2.27, Copyright (C) 1999 - 2007 Digium, Inc. and others. <br>Created by Mark Spencer <<a href="mailto:markster@digium.com">markster@digium.com</a>> <br>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. <br>
This is free software, with components licensed under the GNU General Public <br>License version 2 and other licenses; you are welcome to redistribute it under <br>certain conditions. Type 'show license' for details. <br>
=========================================================================<br>Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2615) <br>Verbosity is at least 21 <br>-- Executing AGI("SIP/cc101-b7a07420", "agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a>") in new stack <br>
-- AGI Script <a href="agi://127.0.0.1:4577/call_log" target="_blank" style="color: rgb(0, 102, 153); ">agi://127.0.0.1:4577/call_log</a> completed, returning 0 <br>-- Executing Dial("SIP/cc101-b7a07420", "SIP/17275691533@sip8||tTor") in new stack <br>
-- Called 17275691533@sip8 <br>-- SIP/sip8-0825f9b0 is making progress passing it to SIP/cc101-b7a07420 <br>-- SIP/sip8-0825f9b0 answered SIP/cc101-b7a07420 <br>== Parsing '/etc/asterisk/manager.conf': Found <br>== Manager 'sendcron' logged on from 127.0.0.1 <br>
== Manager 'sendcron' logged off from 127.0.0.1 <br>== Parsing '/etc/asterisk/manager.conf': Found <br>== Manager 'sendcron' logged on from 127.0.0.1 <br>== Manager 'sendcron' logged off from 127.0.0.1 <br>
== Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc101-b7a07420' <br>-- Executing DeadAGI("SIP/cc101-b7a07420", "agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a>") in new stack <br>
-- AGI Script <a href="agi://127.0.0.1:4577/call_log" target="_blank" style="color: rgb(0, 102, 153); ">agi://127.0.0.1:4577/call_log</a> completed, returning 0 <br>-- Executing DeadAGI("SIP/cc101-b7a07420", "agi://<a href="http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45">127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45</a>)") in new stack <br>
-- AGI Script <a href="agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45" target="_blank" style="color: rgb(0, 102, 153); ">agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----48-----45</a>) completed, returning 0 <br>
vicidialnow*CLI> <br><br><br><br><br><br><br><span class="Apple-style-span" style="color: rgb(0, 0, 0); font-size: 16px; line-height: normal; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; "><span class="postbody" style="font-size: 12px; line-height: 18px; ">sip debug shows below lines: <br>
<br></span><table width="90%" cellspacing="1" cellpadding="3" border="0" align="center"><tbody><tr><td style="font-family: Verdana, Arial, Helvetica, sans-serif; "><span class="genmed" style="font-size: 11px; color: rgb(0, 0, 0); "><b>Quote:</b></span></td>
</tr><tr><td class="quote" style="font-family: Verdana, Arial, Helvetica, sans-serif; font-size: 11px; color: rgb(68, 68, 68); line-height: 125%; background-color: rgb(250, 250, 250); border-top-color: rgb(209, 215, 220); border-right-color: rgb(209, 215, 220); border-bottom-color: rgb(209, 215, 220); border-left-color: rgb(209, 215, 220); border-width: initial; border-style: initial; border-top-style: solid; border-right-style: solid; border-bottom-style: solid; border-left-style: solid; border-left-width: 1px; border-top-width: 1px; border-right-width: 1px; border-bottom-width: 1px; ">
<br><br>--- (12 headers 0 lines) --- <br>Sending to 192.168.0.50 : 12714 (NAT) <br>Transmitting (NAT) to <a href="http://192.168.0.50:12714">192.168.0.50:12714</a>: <br>SIP/2.0 200 OK <br>Via: SIP/2.0/UDP 192.168.0.50:12714;branch=z9hG4bK-d87543-930550325154e53d-1--d87543-;received=192.168.0.50;rport=12714 <br>
From: "cc106"<<a href="mailto:sip%3Acc106@192.168.0.2">sip:cc106@192.168.0.2</a>>;tag=7f1cff22 <br>To: "817275691533"<<a href="mailto:sip%3A817275691533@192.168.0.2">sip:817275691533@192.168.0.2</a>>;tag=as02559696 <br>
Call-ID: NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU. <br>CSeq: 3 BYE <br>User-Agent: Asterisk PBX <br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY <br>Contact: <<a href="mailto:sip%3A817275691533@192.168.0.2">sip:817275691533@192.168.0.2</a>> <br>
Content-Length: 0 <br>X-Asterisk-HangupCause: Normal Clearing <br><br><br><br>--- <br>Scheduling destruction of call '617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx' in 32000 ms <br>set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to <br>
set_destination: set destination to 8.14.xxx.xxx, port 5060 <br>Reliably Transmitting (no NAT) to 8.14.xxx.xxx:5060: <br>BYE sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 <br>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK59c0212a;rport <br>
From: "cc106" <sip:fiddialer@59.xxx.xx.xx>;tag=as3f9466a7 <br>To: <sip:17275691533@8.14.xxx.xxx>;tag=1902000923108720995156225 <br>Call-ID: <a href="mailto:617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx" style="color: rgb(0, 102, 153); ">617ad67d47db8e4a2155fcd51d1089ff@59.xxx.xx.xx</a> <br>
CSeq: 103 BYE <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Content-Length: 0 <br><br><br>--- <br>== Spawn extension (default, 817275691533, 2) exited non-zero on 'SIP/cc106-b7a1a9d0' <br>-- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://<a href="http://127.0.0.1:4577/call_log">127.0.0.1:4577/call_log</a>") in new stack <br>
-- AGI Script <a href="agi://127.0.0.1:4577/call_log" target="_blank" style="color: rgb(0, 102, 153); ">agi://127.0.0.1:4577/call_log</a> completed, returning 0 <br>-- Executing DeadAGI("SIP/cc106-b7a1a9d0", "agi://<a href="http://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12">127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12</a>)") in new stack <br>
-- AGI Script <a href="agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12" target="_blank" style="color: rgb(0, 102, 153); ">agi://127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----12</a>) completed, returning 0 <br>
Destroying call 'NmZlY2E3ZDk0MDVmN2M1MGVkOGJlOTBiYjg5ODIxNTU.' <br>vicidialnow*CLI> <br><-- SIP read from 8.14.xxx.xxx:5060: <br>SIP/2.0 200 OK <br>CSeq: 102 INVITE <br>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK3a111ef4;rport <br>
From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976 <br>Call-ID: <a href="mailto:1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx" style="color: rgb(0, 102, 153); ">1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx</a> <br>
To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221 <br>Contact: <sip:8.14.xxx.xxx:5060;transport=udp> <br>Content-Type: application/sdp <br>Content-Length: 225 <br><br>v=0 <br>o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx <br>
s=VoipSIP <br>i=Audio Session <br>c=IN IP4 8.14.xxx.xxx <br>t=0 0 <br>m=audio 6220 RTP/AVP 18 101 <br>a=rtpmap:18 G729/8000/1 <br>a=rtpmap:101 telephone-event/8000 <br>a=fmtp:101 0-15 <br>a=sendrecv <br><br>--- (9 headers 11 lines) --- <br>
Found RTP audio format 18 <br>Found RTP audio format 101 <br>Peer audio RTP is at port 8.14.xxx.xxx:6220 <br>Found description format G729 <br>Found description format telephone-event <br>Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) <br>
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) <br>list_route: hop: <sip:8.14.xxx.xxx:5060;transport=udp> <br>set_destination: Parsing <sip:8.14.xxx.xxx:5060;transport=udp> for address/port to send to <br>
set_destination: set destination to 8.14.xxx.xxx, port 5060 <br>Transmitting (no NAT) to 8.14.xxx.xxx:5060: <br>ACK sip:8.14.xxx.xxx:5060;transport=udp SIP/2.0 <br>Via: SIP/2.0/UDP 59.xxx.xx.xx:5060;branch=z9hG4bK6eef7893;rport <br>
From: "V0219160007000134649" <sip:fiddialer@59.xxx.xx.xx>;tag=as79fae976 <br>To: <sip:16785588539@8.14.xxx.xxx>;tag=1902000923098720982816221 <br>Contact: <sip:fiddialer@59.xxx.xx.xx> <br>Call-ID: <a href="mailto:1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx" style="color: rgb(0, 102, 153); ">1525f0ef1e787bed51ed1ef119adb1fa@59.xxx.xx.xx</a> <br>
CSeq: 102 ACK <br>User-Agent: Asterisk PBX <br>Max-Forwards: 70 <br>Content-Length: 0 <br><br><br></td></tr></tbody></table></span><br><br><br>Let me know if any other information is required <br><br><br></td></tr></tbody></table>
</span></span></div>