Hi Anusha,

Just emailed you the VXML doc and the audio file.

We are making a call from a SIP phone to 4000@<IP Address>: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

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

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

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 ipv4here 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

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

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

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