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Ryan

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About Ryan

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  1. It would be cool if there was a filesystem-based approach, or something that updates a database everytime a new call comes in or ends. Maybe CDR does this?
  2. Hello, Currently, our site is using Curl to connect to the web admin interface of Pbxnsip, and gets the /ajax.htm?action=call_list&domain=domain.com page. This provides us with a current call list, but it takes almost 4-5 seconds to connect to the web interface. Is there a better way to get a call list, maybe with the new versions of Pbxnsip? It would be nicer to have an instant (< 1s) response to the query. Ryan
  3. Ryan

    MOH RTP Input

    So with the version you linked me to, is RTP shoutcast streaming working perfectly now?
  4. Still nothing? I am going to be forced to move to Asterisk if this isn't working like this week
  5. Ryan

    PHP CSTA example?

    But can't you use CSTA with SOAP?
  6. Ryan

    PHP CSTA example?

    Are there any examples on what to do to run CSTA events? I'd like to do the "MakeCall" action. But I cannot find any examples on doing this.
  7. What whole process? Is this documented somewhere?
  8. OK, this is the documentation I found: http://wiki.snomone.com/index.php?title=CSTA_example But it does not say WHERE I send that data to? Do I POST it to a specific page of the admin interface? Also, how do I know the call ID that is long like that?
  9. Hello, I need to make an API request or SOAP request to the PBX to transfer a current call to another extension. Would I use CSTA? I will be making a web interface via PHP for this; any ideas where to start? Ryan
  10. Ryan

    MOH RTP Input

    Now just waiting on the new build or the VLC instructions, thanks guys
  11. Ryan

    MOH RTP Input

    Can I ask how YOU, Pbxnsip tested the RTP? Because no matter what I try, ffmpeg or vlc, I get a clicking sound in the background and it just sounds terrible.
  12. Ryan

    MOH RTP Input

    --rtp-caching=<integer [0 .. 65535]> RTP de-jitter buffer length (msec) That's all I found in the docs. Googling shows no options for RTP Packet size. Any other ideas?
  13. Ryan

    MOH RTP Input

    Ah, I got it working. I streamed via VLC using: cvlc http://[shoutcastip]:[shoutcastport] --loop --norm-max-level=5 --sout='#transcode{acodec=ulaw,samplerate=8000,channels=1,ab=16}:rtp{dst=[PBXNSIP IP],port-audio=[PBXNSIP MOH PORT]]' This is streaming an MP3 shoutcast radio stream to the RTP on Pbxnsip. It seems to be streaming when I put myself on hold. It is however very choppy, chopping about every second or 2. I will try and run VLC on the same box and see if that improves the choppiness. Any suggestions for streaming RTP with less choppiness are welcome.
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