No audio on call forward after upgrade from Asterisk 1.4 to 1.6

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Asterisk Users 5 Comments

I have a client that has an Ubuntu x64 based Asterisk server with and ISDN Dahdi interface and about 25 SIP handsets. Everything was working fine in Asterisk 1.4 and now after migrating the config to Asterisk 1.6.2.5 I have one single issue that I can’t explain.

I have an extension that if you call it, it will play a sound file and hangup. Pretty simple stuff. Below is the extensions.conf entry for this extension.

exten => 849,1,Playback(custom/ceh-meetingmsg)
exten => 849,n,Hangup

5 thoughts on - No audio on call forward after upgrade from Asterisk 1.4 to 1.6

  • Hi Alex,

    I’m new to this list, but I had this problem too, and I solved it looking at
    the codecs the sip handsets use, and then I converted the voice prompts to
    that codec just like Philipp said..

    Ondrej

  • exten => 849,1,Progress()
    exten => 849,n,Playback(custom/ceh-meetingmsg)
    exten => 849,n,Hangup

  • Hi Paul,

    I tried adding Progress() to no avail. I still get no audio and below is what comes up in the console.

  • Try moving Progress() before the Dial(). If you Answer() the channel,
    do you have the same problem?

  • Hi Paul,

    No cigar unfortunately. I also tried encoding the message as gsm, ulaw and alaw with no success.

    The ISDN interface is alaw and the SIP phones I was testing with are definately alaw.

    Not sure what to do from here. I might just need to bypass the issue using some alternate way to put the message in front of the inbound dialplan logic on some condition.

    aF

    href=”mailto:paul.belanger@polybeacon.com”>paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)