VXML SIP Transferaudio

Version 1
    This document was generated from CDN thread

    Created by: Heimo Stieg on 08-04-2013 09:28:19 AM
    Hello, I've created a simple script to play an audio file and redirect it to the operator(CUCM).
    It is working fine on PSTN calls, but as soon as the source is a SIP call, then the call will be transfered without audio.  
     
    Has the router be configured in a special way to use early-media with sip in a vxml application? Or am I missing something in the vxml script?
     
    Kind Regards
    Heimo 

    Subject: RE: VXML SIP Transferaudio
    Replied by: Yaw-Ming Chen on 08-04-2013 11:11:24 AM
    Can you plrase attached the dial-peer configuration for this vxml service ?
    Thanks !

    Subject: RE: VXML SIP Transferaudio
    Replied by: Heimo Stieg on 09-04-2013 02:44:27 AM
    The pots dial peer has the same configuration:
    dial-peer voice 52 voip
     description "SIP Test"
     service tvmsip
     incoming called-number 4321T


    The sip dial-peer:
    voice class codec 10
     codec preference 1 g711ulaw
     codec preference 2 g729r8
     codec preference 3 g711alaw


    dial-peer voice 51 voip
     destination-pattern [1-8]T
     voice-class codec 10
     session protocol sipv2
     session target ipv4:<IP>
     dtmf-relay rtp-nte


    Subject: RE: VXML SIP Transferaudio
    Replied by: Heimo Stieg on 09-04-2013 04:02:34 AM
    Hi Raghavendra,  
    you can find the log in the attachement.  
    Thanks =) 

    Subject: RE: VXML SIP Transferaudio
    Replied by: Raghavendra Gutty Veeranagappa on 09-04-2013 03:49:13 AM
    Hi Heimo,
    could you please send us the logs by enabling below debugs.
    debug voip app
    Thanks,
    Raghavendra

    Subject: RE: VXML SIP Transferaudio
    Replied by: Raghavendra Gutty Veeranagappa on 09-04-2013 04:02:07 AM
    Hi Heimo,
    please try to configure "codec g711ulaw" to your dial-peer 52.audio files also should be same codec.
    Thanks,
    Raghavendra

    Subject: RE: VXML SIP Transferaudio
    Replied by: Raghavendra Gutty Veeranagappa on 09-04-2013 04:24:01 AM
    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

    Subject: RE: VXML SIP Transferaudio
    Replied by: Heimo Stieg on 09-04-2013 04:27:14 AM
    Thank you Raghavendra
    the missing codec in the dial-peer was the problem.
     
    Regards
    Heimo