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Hi there, I know this question is kind of stupid. But, I don't know anywhere else to ask. I've received some answers when I asked about the need of having a zaptel interface to make the meetme application work, that said that it was better to have a real hardware then the zaptelrtc software modules. So, my question is, would any of the following cards work with the meetme

Hi there, I would like to communicate H323 IP phones with SIP phones. My H323 phones are registered to a gnugk GK, and the SIP phones are registered to a asterisk SIP proxy. I could not create a dialplan that works. Inside my extensions.conf file I created the following two entrances: exten => 4,1,Dial(SIP/4) exten => 5,1,Dial(SIP/5) This allows SIP phones call each other.

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

-This is my 'sip.conf' file: ;************************************************************* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow

Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

Hi there, I am trying to register asterisk to a Lucent's MVAM GK. It is not registering to the GK with both h323 channels(chan_h323 and oh323). The problem is that if I set the GK(in asterisk) through the GK's IP, the GK answers with GCF without it's GK ID, and after that it does not answer to the RRQ, because the RRQ message has no GK ID. And when I set the GK through

Hi there, I have the following scenario: I have a GK that controls my H323 GWs. And, I am needing an IVR royalty-free system. More than an IVR, it's only something that should play a sound file depending on the called digits. So, I wanted to use asterisk as my IVR. So, I've installed asterisk with the chan_h323 module and used the Playback comand in my extensions.conf

Hi there, I want to have all my sip.conf data inside a DataBase, so that my asterisk admintration system would be through a Web Interface connected to the DB. Is there any way to put the sip.conf file in a Data Base and then to read from it, in such a way that the sip.conf file would have some line that points to the DataBase? I have seen wiki's page

Hi there, I tried to install appradius with asterisk. The appradius instalation finishes succesfully, but the problem is that I can't find the app_radius.so and cdr_radius.so files that the installation indicates to copy to the /usr/lib/asterisk/modules directory(http://appradius.minitelecom.org/docs/cdr_radius.html). Was somebody able to install this module? Does it work? I've