<?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_thread?p_l_id=&amp;threadId=6238748" />
  <subtitle>RE: Direct SIP call</subtitle>
  <id>http://developer.cisco.com/c/message_boards/find_thread?p_l_id=&amp;threadId=6238748</id>
  <updated>2013-05-25T00:58:02Z</updated>
  <dc:date>2013-05-25T00:58:02Z</dc:date>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6455451" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6455451</id>
    <updated>2012-09-07T10:59:45Z</updated>
    <published>2012-09-07T10:59:45Z</published>
    <summary type="html">Hi Raghavendra

Thanks for spending time to see the logs.

I tried the 'media flow-through' command, but interestingly, even though I configure the command in the dial-peer, when I do a show run | s dial, it does not appear in the dial-peer ?

I made a call after that but still cant hear the audio :-(

Regards
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-07T10:59:45Z</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=&amp;messageId=6453063" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&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=&amp;messageId=6433243" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&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=&amp;messageId=6433211" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6433211</id>
    <updated>2012-09-06T12:26:54Z</updated>
    <published>2012-09-06T12:26:54Z</published>
    <summary type="html">Hi Raghavendra,

Actually, we have been working with TAC in the past to figure out these issues when we were setting up the other interface (with E1 and not direct-SIP) but got to know from them that TAC can only be used for break-and-fix issues in a running deployment and it would not be able to cover issues for setting up a new deployment :-(

This is a bit time-critical for us, so we wanted to get a headway into this soon enough. Any help from your team would be great for us.

(Also, if you find anything significant in the logs, please let us know)

Regards,
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-06T12:26:54Z</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=&amp;messageId=6433185" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&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=&amp;messageId=6433157" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6433157</id>
    <updated>2012-09-06T11:53:35Z</updated>
    <published>2012-09-06T11:53:35Z</published>
    <summary type="html">Hi Raghavendra,

The audio file is not playing after the upgrade :-( But I am thinking that the issue is with the dial-peers since I guess its going into a recursive loop.

Sending you the logs on email. Thanks again.

Regards,
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-06T11:53:35Z</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=&amp;messageId=6433115" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&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>
  <entry>
    <title>RE: Direct SIP call</title>
    <link rel="alternate" href="http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6433011" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6433011</id>
    <updated>2012-09-06T10:47:37Z</updated>
    <published>2012-09-06T10:47:37Z</published>
    <summary type="html">Hi Raghavendra/Anusha,

We have now upgraded the IOS to Version 15.1(4)M3, RELEASE SOFTWARE (fc1) (We had to make a couple of changes in our vxml and the config to accomodate the features of the new IOS (like the Toll-Fraud Prevention Feature in IOS Release 15.1(2)T) etc. but the config is now set and we can make SIP calls.

However, when we call, we get the following error in the logs:

000132: *Sep  6 09:12:01.259: //-1/xxxxxxxxxxxx/CCAPI/cc_set_post_tagdata:
   CALL_ERROR; Avlist Set Is Failed
000133: *Sep  6 09:12:01.259: //-1/C07ED4A98002/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=anupam
   ----- ccCallInfo IE subfields -----
   cisco-ani=anupam
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=4000
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=-1
   cisco-rdnsi=-1
   cisco-redirectreason=-1   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0



We are wondering if it is still an issue with our dial-peers. Let me re-describe our situation here - We are using the CIsco IOS for VXML interpretation and not using CVP. The VXML is hosted on our own app server externally. The call is directly coming in as a SIP Call (without any E1 interface) and it invokes the VXML from the app server. On a separate instance of 2851 router, we have tested this integration with a E1 interface. Now we want to do the same thing without E1 and using VOIP-only.

Here is our dial-peer:

dial-peer voice 2 voip
 service myapp
 destination-pattern 4000
 session protocol sipv2
 session target ipv4:(here we specify the IP of the same router back to itself)
 incoming called-number 4000
 voice-class codec 1
 dtmf-relay sip-notify

I think its going in a loop because session-target is pointing back to the same router on which this dial-peer is defined?

Thanks
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-06T10:47:37Z</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=&amp;messageId=6414298" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6414298</id>
    <updated>2012-09-04T12:12:39Z</updated>
    <published>2012-09-04T12:12:39Z</published>
    <summary type="html">Thanks Raghavendra.

We have turned off the prompt streaming now but looks like the issue still remains, we still cant hear the audio. 

We are trying to upgrade the IOS. Will keep you posted. Thanks.

Regards
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-04T12:12:39Z</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=&amp;messageId=6414249" />
    <author>
      <name>Raghavendra Gutty Veeranagappa</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6414249</id>
    <updated>2012-09-04T11:41:04Z</updated>
    <published>2012-09-04T11:41:04Z</published>
    <summary type="html">Hi Anupam,

you can turn off prompt streaming with below command.

 no ivr prompt streamed 

you can find the command in below link
http://www.cisco.com/en/US/docs/ios/12_3t/voice/command/reference/vrht_i2_ps5207_TSD_Products_Command_Reference_Chapter.html#wp1102051

Thanks,
Raghavendra</summary>
    <dc:creator>Raghavendra Gutty Veeranagappa</dc:creator>
    <dc:date>2012-09-04T11:41:04Z</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=&amp;messageId=6414098" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6414098</id>
    <updated>2012-09-04T10:20:30Z</updated>
    <published>2012-09-04T10:20:30Z</published>
    <summary type="html">Hi Anusha

(I posted a reply earlier but looks like it got overridden by a an old post somehow)

Thanks a bunch for testing this inhouse and pointing us to the Defect #. We are now trying to upgrade the IOS.

Also, we tried to search for a command to turn off the prompt streaming mode (as mentioned in the workaround) but could find one instance at http://www.cisco.com/en/US/docs/ios/11_0/access/connection/guide/xuserif.html where the command is like 'resume /nostream'. Not sure if it is for the same purpose. Can you help us with it?

Thanks
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-04T10:20:30Z</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=&amp;messageId=6413907" />
    <author>
      <name>Anusha Kannappan</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6413907</id>
    <updated>2012-09-04T09:03:14Z</updated>
    <published>2012-09-04T09:03:14Z</published>
    <summary type="html">Hi Anupam,

Actually we had tried with SIP and SCCP phones and were able to hear the audio file being played in our lab. Please check whether you are hitting the following Defect CSCsl61416. Try to upgrade the IOS version or work around specified over there and see whether it resolves your issue.

Thanks,
Anusha</summary>
    <dc:creator>Anusha Kannappan</dc:creator>
    <dc:date>2012-09-04T09:03:14Z</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=&amp;messageId=6413760" />
    <author>
      <name>Anusha Kannappan</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6413760</id>
    <updated>2012-09-04T08:06:55Z</updated>
    <published>2012-09-04T08:06:55Z</published>
    <summary type="html">Hi Anupam,

Thanks for sharing the docs, we will test this in our lab and revert back to you.

Thanks,
Anusha</summary>
    <dc:creator>Anusha Kannappan</dc:creator>
    <dc:date>2012-09-04T08:06:55Z</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=&amp;messageId=6413720" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6413720</id>
    <updated>2012-09-04T07:41:20Z</updated>
    <published>2012-09-04T07:41:20Z</published>
    <summary type="html">Hi Anusha,

Just emailed you the VXML doc and the audio file.

We are making a call from a SIP phone to 4000@&lt;IP Address&gt;:5060.

Also, we just observed that after about 10-11 seconds of the call, we hear a short disturbance for 1 second (as if the whole audio is being played very very fast in one second)

Thanks
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-04T07:41:20Z</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=&amp;messageId=6413678" />
    <author>
      <name>Anusha Kannappan</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6413678</id>
    <updated>2012-09-04T07:28:51Z</updated>
    <published>2012-09-04T07:28:51Z</published>
    <summary type="html">Hi Anupam,

Is it possible to share us the VXML doc along with the audio files to test the same in our local lab. Meanwhile if you have valid developer support contract id please raise a developer support case for the same to track this issue. Also mention the clear call flow that you are trying in your setup.

Thanks,
Anusha</summary>
    <dc:creator>Anusha Kannappan</dc:creator>
    <dc:date>2012-09-04T07:28:51Z</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=&amp;messageId=6410284" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6410284</id>
    <updated>2012-09-03T10:44:11Z</updated>
    <published>2012-09-03T10:44:11Z</published>
    <summary type="html">Hi Anusha,

I added the following 

session protocol sipv2 
dtmf-relay sip-notify

in the dial-peer but the problem still persists :-( Cant hear anything on the call. 

The dial-peer now looks like the following:

dial-peer voice 2 voip
 service myapp
 destination-pattern 4000
 voice-class codec 1
 session protocol sipv2
 session target ipv4:203.122.28.219
 incoming called-number 4000
 dtmf-relay sip-notify

Thanks
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-03T10:44:11Z</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=&amp;messageId=6411533" />
    <author>
      <name>Anusha Kannappan</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6411533</id>
    <updated>2012-09-03T10:35:37Z</updated>
    <published>2012-09-03T10:35:37Z</published>
    <summary type="html">Hi Anupam,

Try adding the following cli to the 'dial-peer voice 2 voip' and see whether it resolves your issue

session protocol sipv2

Thanks,
Anusha</summary>
    <dc:creator>Anusha Kannappan</dc:creator>
    <dc:date>2012-09-03T10:35:37Z</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=&amp;messageId=6410185" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6410185</id>
    <updated>2012-09-03T09:25:31Z</updated>
    <published>2012-09-03T09:25:31Z</published>
    <summary type="html">Hi Anushka,

Thanks for helping out on this. Sent you the logs and configuration on email.

Regards
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-03T09:25:31Z</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=&amp;messageId=6411364" />
    <author>
      <name>Anusha Kannappan</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6411364</id>
    <updated>2012-09-03T08:16:38Z</updated>
    <published>2012-09-03T08:16:38Z</published>
    <summary type="html">Hi Anupam,

Could you please share us the running config along with the logs with the following debugs enabled.

deb voip app
deb ccsip all

Let me see whether I could trace anything out of this !

Thanks,
Anusha</summary>
    <dc:creator>Anusha Kannappan</dc:creator>
    <dc:date>2012-09-03T08:16:38Z</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=&amp;messageId=6410049" />
    <author>
      <name>Anupam Jain</name>
    </author>
    <id>http://developer.cisco.com/c/message_boards/find_message?p_l_id=&amp;messageId=6410049</id>
    <updated>2012-09-03T07:53:48Z</updated>
    <published>2012-09-03T07:53:48Z</published>
    <summary type="html">Hi Anusha,

Thanks again for your prompt response.

This is the IOS version output from 'show ver':
Cisco IOS Software, 2800 Software (C2800NM-ADVIPSERVICESK9-M), Version 12.4(15)T8, RELEASE SOFTWARE (fc3)

Also, in the meanwhile, I added the following line to the dial-peer:
voice-class codec 1

and the call seems to be going through now (as in, it does not throw a 488)

But still, we are not able to hear the audio. All we are doing in the VXML is playing a wav file and exiting. (This vxml is constructed fine since we have already tested it with another instance of a Cisco router sometime back)

From the logs, it seems that the file did play and the application exited properly but we cant hear anything. I suspect it may still be a config issue but couldnt figure it out as yet.

Thanks
Anupam</summary>
    <dc:creator>Anupam Jain</dc:creator>
    <dc:date>2012-09-03T07:53:48Z</dc:date>
  </entry>
</feed>

