Jump to content
Vodia PBX forum

Ryan

Members
  • Content Count

    73
  • Joined

  • Last visited

Everything posted by Ryan

  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.
  14. Ryan

    MOH RTP Input

    It starts listening on the port only when a call is placed on hold? It's not always listening? That is odd....How can I have a client connect and stream to a nonexistent port? I need to set something up and have it streaming to that port.
  15. Ryan

    MOH RTP Input

    This is kind of urgent and a customer needs a reply soon; is this a bug in the system since it's not listening on the port I specified? Or do I have to do something else after adding that MOH?
  16. Ryan

    MOH RTP Input

    As stated, when I saved that music on hold item, and restart the PBX service, and run `netstat`, it is NOT listening on the port I specified. I am going to be using VLC to stream data to the specified port. But I cannot stream to that port until the PBX is listening on the port.
  17. Ryan

    MOH RTP Input

    Version is the latest CentOS Linux pbxctrl-centos5-4.2.1.4025
  18. Ryan

    MOH RTP Input

    When selecting a new Music On Hold source, there is an option for "RTP Stream". It asks for a port number; does the PBX actually listen on this port? I have tried entering an unused port here, and restarting the server, and it does not listen on that port. Am I doing something wrong?
  19. ? Can you explain what you just linked me to? What does "should have the same functionality as the pbxnsip." mean? Should I try one of these modules? Also, in the Modules section, the MOH thing is for Windows only. I am using Linux.
  20. Hello, We need to have the Music on Hold play a Shoutcast stream. I know Pbxnsip expects a specific WAV format for the files, and I am wondering if anyone knows how to make this work in pbxnsip, even if it's using other tools to pipe to a .wav file. I know Asterisk has this built-in, and you can just enter mpg123 and a stream name. I hope pbxnsip has some kind of solution for this! Regards, Ryan
  21. How about this. Inside /pbx/extensions/[ext id].xml, there is a field: <vm_indicator>1/0 (0/0)</vm_indicator> I notice on my full mailbox, it said "100/0 (0/7)". Now, after recreating the account, I have 1 voicemail, it shows "1/0 (0,0)". Could I simply change the "<vm_indicator>1/0 (0/0)</vm_indicator>" to "<vm_indicator>0/0 (0/0)</vm_indicator>" each time my script runs? Is this the only instance that needs to change?
  22. Hello, I have a script that runs every 15 minutes, checks the /pbx/messages dir, and if it sees a new file, updates my sql database with info on it. It then moves the file from /pbx/messages/[id].xml to my own dir, and moves the corresponding recording from the /pbx/recordings dir. As far as I know this is all I need to do, but the voicemail box still thinks there are 100 messages (I have max set to 100). How can I clear the voicemail box or make it show 0/100 after I move the voicemails? Ryan
  23. When I hit the 'ajax.htm' page (which I'm using to get an active call list), it gives me just my auto attendant. But the admin interface will show the auto attendant AND the connected agent ID, such as: 600 (602) Where 600 is the AA and 602 is the agent ID. How can I manipulate the ajax.htm to give me what the admin interface gives me?
×
×
  • Create New...