Ryan
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Posts posted by Ryan
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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
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So with the version you linked me to, is RTP shoutcast streaming working perfectly now?
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CentOS 5, and 32bit.
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Still nothing? I am going to be forced to move to Asterisk if this isn't working like this week
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But can't you use CSTA with SOAP?
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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.
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What whole process? Is this documented somewhere?
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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?
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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
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Now just waiting on the new build or the VLC instructions, thanks guys
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Glad to hear it!
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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.
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--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?
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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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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.
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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?
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Yes, the pbx actually listen to RTP ports, how are you streaming the media?
http://wiki.snomone.com/index.php?title=Music_on_Hold_Sources
"Important: Be sure to specify the port on which the system should listen for RTP input (e.g., 42000). This port must be available on the system. If you change the setting, you might need to restart the system service so that the change takes effect"
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.
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Version is the latest CentOS Linux pbxctrl-centos5-4.2.1.4025
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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?
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You can download the app here
http://wiki.snomone.com/index.php?title=Downloads
should have the same functionality as the pbxnsip.
http://downloads.snom.net/snomONE/docs/snomONE_online_book.pdf Check page 385
If you are running a pbxnsip version nothing has really change in this area.
? 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.
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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
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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?
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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
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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?
Best method for currently active call list
in General Setup
Posted
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?