search for: kutman

...fter going through asterisk it seems like asterisk adds a codec because the SDP body now contains the codecs 0 and 3. I believe the problem has something to do with this but I am not sure why it would work one way but not the other. Any help would be greatly appreciated. Thanks very much, Denis Kutman

Hello, Is it possible to print the Asterisk message logs to a file, or is this already done? By message logs I mean the display that shows up on the asterisk server when a call is made from one user to another. I believe if the verbosity is high, it can show what parts of the extension.conf file that it uses when making the call. I am trying to use two Jain-sip-applet-phones, connected through

...I believe Asterisk does accept subscription requests, but for some reason it doesn't like this one. I will soon start to experiment with the source code. Thanks, Denis -----Original Message----- From: Gerald A [mailto:geraldablists at gmail.com] Sent: Monday, August 27, 2007 9:30 AM To: Kutman DK at ADM(Mat) DAEPM(R&CS)@Ottawa-Hull Subject: Re: [asterisk-users] Can't create audio conversation between softphonesthrough Asterisk Hi, On 8/27/07, Kutman.DK at forces.gc.ca < Kutman.DK at forces.gc.ca > wrote: Thanks for the reply. I have a small LAN network which I have...

Hello, I have been reading up on the capabilities of the Asterisk-Java library. I believe that this library can act as an interface between a Java GUI(custom softphone) and the Asterisk server. Seems like the Live API would be easiest to use to make the connection to the Asterisk server and to set-up calls. One thing I am not sure about is how to transmit the audio streams between users'

...SIP/201 context=from-internal canreinvite=no callerid=device <201> [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device <202> Thanks very much, Denis Kutman

...re is an rtp.conf file which outlines the ports available for rtp transfer. How is the actual RTP transfer between users completed through the Asterisk server? I am looking to transmit live audio between the users through the Asterisk server once the call is connected. Thanks in advance, Denis Kutman

Hello, I have a small LAN network connected through an Asterisk Server (Trixbox). I was looking to create my own custom made softphones, and I have been looking into how to transmit and receive via RTP. Would anyone know how the Asterisk RTP bridging works, and if there is any documentation on it? How is the RTP stream routed through the Asterisk server? Do I just give it the endpoints and

...record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 context=from-internal canreinvite=no callerid=device <202> Would anyone know where I am going wrong? Is it possible to remove this authentication requirement? Thanks in advance, Denis Kutman

Hello, I have a small LAN network where I am running a Jain-Sip softphone on two user pc's. These softphones are connected through Asterisk(Trixbox). Although the phones do work in providing an audio conversation, there is a long delay(about 20 seconds) in the initial RTP session setup. I have tried a few values for the buffer length including setting it to zero. I assumed this would

Hello, I was wondering if it is possible within Asterisk to be in many meetme conferences at the same time. This would be sort of broadcasting over all the conferences at once. Thanks, Denis

Hello, I have two user machines, each with a jain-sip-applet-phone installed on it. I use the following process to try to make a call: 1. Register each phone with the Asterisk server (working). 2. Add a contact in each phone which is the other user. (Get a "489 Bad Event" SIP error shown below in red) 202 at 192.168.1.252 has been added to your contacts. null send request:

Hello, I am trying to transmit and receive sound over IP using the Java Media Framework(JMF) RTP. I was wondering if its possible to create an RTP Stream from my own computer and assign it to a URL. If anyone knows how I would do this, could they point me to some instructions or an example. So far, I have some sample code which compiles and creates the player, but it can't seem to realize