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Hi all, I'm trying to set up a test case for an ISP featuring an asterisk server and a couple of linksys rt31p2-na routers registering on it. Instead of using dsl lines, i'm trying to plug the * server and the routers on a cisco switch, just to test their functionality. I have created a vlan and a subnet on the switch and set up the ip addresses of the routers in that subnet. When i plug

Hi all, I have installed a TDM22B on an IBM xSeries 220 with FedoraCore 3 and setup asterisk cvs head to work properly. A few days ago, I tried to update the system kernel to 2.6.12 from 2.6.9 and also to change to asterisk stable 1.0.9. After compiling zaptel and asterisk, i loaded zaptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file

Hi all, I have an IAX connection between two asterisk servers and i'm looking for a way to cut down on the needed bandwidth. Both voice and fax calls pass through the channel so it is currently configured to use g.711. Could it be possible to select the codec based on the call's prefix so that g.711 will be used for fax calls and g.729 for voice? Dionisis -------------- next part

Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only strange thing is the "got event On hook" (i guess..). I am thinking that it is a problem

Hello to all, I post again (last time subject: Line drops strange problem(got event On hook) because i have caught in debug a situation where i get a call and the line drops and i get a call from the same caller and the line works well and the call normally closes by both parties. The only differences i find are underlined. If someone can understand the reason why the line drops from the debug