<?xml version="1.0" encoding="UTF-8"?>
<feed xmlns="http://www.w3.org/2005/Atom" xmlns:dc="http://purl.org/dc/elements/1.1/">
  <title>RE: Direct SIP call</title>
  <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_category?p_l_id=148613&amp;mbCategoryId=0" />
  <subtitle>RE: Direct SIP call</subtitle>
  <id>http://developer.cisco.com/c/message_boards/find_category?p_l_id=148613&amp;mbCategoryId=0</id>
  <updated>2013-05-21T05:04:12Z</updated>
  <dc:date>2013-05-21T05:04:12Z</dc:date>
  <entry>
    <title>RE: Only the first 2 prompts play</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15316475" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15316475</id>
    <updated>2013-05-16T06:55:21Z</updated>
    <published>2013-05-16T06:55:21Z</published>
    <summary type="html">Hi Mark,
thanks for sharing the info, i don't find any issue with your config or with the script.
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-05-16T06:55:21Z</dc:date>
  </entry>
  <entry>
    <title>RE: Only the first 2 prompts play</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15275682" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15275682</id>
    <updated>2013-05-15T12:38:25Z</updated>
    <published>2013-05-15T12:38:25Z</published>
    <summary type="html">Hi Mark,
Which IOS version your using, is it possible to share "sh version" and running config.
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-05-15T12:38:25Z</dc:date>
  </entry>
  <entry>
    <title>RE: Only the first 2 prompts play</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15274322" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15274322</id>
    <updated>2013-05-15T12:00:28Z</updated>
    <published>2013-05-15T12:00:28Z</published>
    <summary type="html">Hi Mark,
please try below command and see if it plays all  prompts.
media play leg_incoming 1.au %s10 2.au %s10 3.au %s10 4.au
Thanks,
Raghavendra
 </summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-05-15T12:00:28Z</dc:date>
  </entry>
  <entry>
    <title>RE: Only the first 2 prompts play</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15146565" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15146565</id>
    <updated>2013-05-10T12:22:40Z</updated>
    <published>2013-05-10T12:22:40Z</published>
    <summary type="html">Hi Mark,
Thanks sharing the logs, please try with below command also try to play prompts from flash instead of ftp server.
ivr prompt memory 128
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-05-10T12:22:40Z</dc:date>
  </entry>
  <entry>
    <title>RE: Only the first 2 prompts play</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15142746" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=15142746</id>
    <updated>2013-05-10T09:32:33Z</updated>
    <published>2013-05-10T09:32:33Z</published>
    <summary type="html">Hi Mark,
 
please send us the below debugs.
 
debug voip app
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-05-10T09:32:33Z</dc:date>
  </entry>
  <entry>
    <title>RE: Error no resource when using TTS</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14656414" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14656414</id>
    <updated>2013-04-25T13:32:33Z</updated>
    <published>2013-04-25T13:32:33Z</published>
    <summary type="html">Hi Grant,
 
i dont think there is an issue with VXML, could you please contact TAC they might help you.
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-04-25T13:32:33Z</dc:date>
  </entry>
  <entry>
    <title>RE: VXML SIP Transferaudio</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035908" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035908</id>
    <updated>2013-04-09T09:24:01Z</updated>
    <published>2013-04-09T09:24:01Z</published>
    <summary type="html">Hi Heimo,
from the logs it shows that play audio failed because of codec mismatch, please configure "codec g711ulaw" to your dial-peer 52.
 
Apr  9 08:54:20.737: //34//MSM :/ms_asDone_buginf: Stream Association Failed: Requested codec=0x5=g711ulaw, Negotiated codec=0x10=g729r8
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-04-09T09:24:01Z</dc:date>
  </entry>
  <entry>
    <title>RE: VXML SIP Transferaudio</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035662" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035662</id>
    <updated>2013-04-09T09:02:07Z</updated>
    <published>2013-04-09T09:02:07Z</published>
    <summary type="html">Hi Heimo,
please try to configure "codec g711ulaw" to your dial-peer 52.audio files also should be same codec.
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-04-09T09:02:07Z</dc:date>
  </entry>
  <entry>
    <title>RE: VXML SIP Transferaudio</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035429" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=14035429</id>
    <updated>2013-04-09T08:49:13Z</updated>
    <published>2013-04-09T08:49:13Z</published>
    <summary type="html">Hi Heimo,
could you please send us the logs by enabling below debugs.
debug voip app
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-04-09T08:49:13Z</dc:date>
  </entry>
  <entry>
    <title>RE: Actual reason for call disconnection</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=13759449" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=13759449</id>
    <updated>2013-04-02T10:51:06Z</updated>
    <published>2013-04-02T10:51:06Z</published>
    <summary type="html">Hi Mark,
 
please try to use "infotag get evt_last_disconnect_cause" command. for more information refer TCL IVR programming guide.
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-04-02T10:51:06Z</dc:date>
  </entry>
  <entry>
    <title>RE: Annoucement on Local gateway for IVR Calls</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=12690413" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=12690413</id>
    <updated>2013-03-06T09:34:19Z</updated>
    <published>2013-03-06T09:34:19Z</published>
    <summary type="html">Hi Hariharan,
 
please post CVP queries in below forum
http://developer.cisco.com/web/cvp/community
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2013-03-06T09:34:19Z</dc:date>
  </entry>
  <entry>
    <title>RE: Calls to be in Queue on Operator phone till she picks up the call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=9508441" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=9508441</id>
    <updated>2012-12-18T08:58:03Z</updated>
    <published>2012-12-18T08:58:03Z</published>
    <summary type="html">Hi Hari,

yes you can use this script, When an AA is configured for drop-through mode, the AA sends incoming calls directly to a call queue without providing menu choices to callers. Once in the queue, a caller hears ringback if an agent is available or music on hold (MOH) if all agents are busy. If a prompt for drop-through mode is configured, the caller hears the prompt before being sent to the queue as described. The drop-through prompt is simply a greeting to callers; it might say "Thank you for calling XYZ, Inc. An agent will be with you shortly." 

 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-12-18T08:58:03Z</dc:date>
  </entry>
  <entry>
    <title>RE: multiple language IVR</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=9251808" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=9251808</id>
    <updated>2012-12-11T07:18:46Z</updated>
    <published>2012-12-11T07:18:46Z</published>
    <summary type="html">Hi,
 
"infotag set med_language [index | prefix prefix]" sets the current active language for media playout. you can use this command to change the language. please refer programming guide for more information.
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-12-11T07:18:46Z</dc:date>
  </entry>
  <entry>
    <title>RE: Can the B-ACD queue cope with PSTN numbers in the queue's hunt-group?</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=8586445" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=8586445</id>
    <updated>2012-11-15T13:34:39Z</updated>
    <published>2012-11-15T13:34:39Z</published>
    <summary type="html">Hi David,
If a member of the hunt group is available 
to take a call, the call is connected. If no member of the hunt group is 
available, the call remains in the call queue for that hunt group. Note that a 
call queue, although dedicated to a particular hunt group, is managed by the 
B-ACD call-queue script and not by the hunt group itself.
Call-queue parameters are configurable. You can specify the amount of time that passes between hunt-group retries and the maximum amount of time that a call can be held in queue.
for more information refer below link
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1015759
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-11-15T13:34:39Z</dc:date>
  </entry>
  <entry>
    <title>RE: TCL CPU usage</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=8583487" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=8583487</id>
    <updated>2012-11-15T12:01:53Z</updated>
    <published>2012-11-15T12:01:53Z</published>
    <summary type="html">Hi Mark,
 
For the higher number of concurrent calls requires more memory, try playing less media files so that less memory is being used.
 
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-11-15T12:01:53Z</dc:date>
  </entry>
  <entry>
    <title>RE: Transfer incoming call before call connect</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=7304072" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=7304072</id>
    <updated>2012-10-03T12:03:24Z</updated>
    <published>2012-10-03T12:03:24Z</published>
    <summary type="html">Hi Grant,
when destination does not answer TCL script will recieve ls_002 status.
ls_002 -- The call setup timed out (meaning that the destination phone was alerting, but no one answered). The limit of this timeout can be specified in the leg setup command.
Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-10-03T12:03:24Z</dc:date>
  </entry>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6453063" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6453063</id>
    <updated>2012-09-07T10:40:31Z</updated>
    <published>2012-09-07T10:40:31Z</published>
    <summary type="html">Hi Anupam,

from the logs i didn't find any issue, please try to configure below command to dial peer.

dial-peer voice 2 voip
  media  flow-through

Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-09-07T10:40:31Z</dc:date>
  </entry>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433243" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433243</id>
    <updated>2012-09-06T12:40:33Z</updated>
    <published>2012-09-06T12:40:33Z</published>
    <summary type="html">Hi Anupam,

we will not be able to help much since issue with your configuration, any way you can send logs to developer-support@cisco.com , if you don't want share logs here.

Thanks,
Ragahvendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-09-06T12:40:33Z</dc:date>
  </entry>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433185" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433185</id>
    <updated>2012-09-06T12:19:53Z</updated>
    <published>2012-09-06T12:19:53Z</published>
    <summary type="html">Hi Anupam,

since it is configuration issue you can raise TAC case, they will help you on this.

Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-09-06T12:19:53Z</dc:date>
  </entry>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433115" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=148613&amp;messageId=6433115</id>
    <updated>2012-09-06T11:39:24Z</updated>
    <published>2012-09-06T11:39:24Z</published>
    <summary type="html">Hi Anupam,

is audio file is playing now after upgrade,  please send us the logs with below debugs.

deb voip app
deb ccsip all

Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-09-06T11:39:24Z</dc:date>
  </entry>
</feed>

