similar to: Sending live audio in Asterisk

Hi, In the early stages of deciding how to try and develop this environment, I looked at all the protocols that could be used. SIP was chosen just because it seemed to me that it was the most widely used protocol. I believe IAX is a new protocol with a little less documentation and examples. The good thing about this Jain-sip-phone is that it saves a lot of time since many of the important

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'

On 03/11/03 00:25, Mark Spencer wrote: > As a side note, I strongly would like to see someone implement a > client using libiax2 which implements IAX2 instead of the (now > obsolescent) IAX version 1. I'm implementing a Java-based IVR server (and yes, I know Asterisk does IVR, and no, it's not flexible enough to do what I want and no, it doesn't integrate well with the Java

Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Would anyone know why

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

I'm currently involved in a project where in we need to record the audio from a mic encode with vorbis, then put it in Ogg container, and then make it as RTP packet and transmit over Darwin Streaming Server. Till now we were using JMF with available codec's, since JMF handles the RTP part life was pretty much easy using it. Now we want to use Ogg Vorbis because of its adv's. So in

I'm currently involved in a project where in we need to record the audio from a mic encode with vorbis, then put it in Ogg container, and then make it as RTP packet and transmit over Darwin Streaming Server. Till now we were using JMF with available codec's, since JMF handles the RTP part life was pretty much easy using it. Now we want to use Ogg Vorbis because of its adv's. So in

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 have the Jain-Sip-Applet-Phone installed on two machines in a small LAN network. These machines are connected through an Asterisk Server (Using Trixbox). I run the phone as an application on both machines through Eclipse and I am able to log on as a user with one of the extensions that I use within Asterisk on each machine (extensions 201 and 202 in this case). When I try to add a

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

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

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

So, I was running my full backup last night, backing up my main data pool zp1, and it seems to have hung. Any suggestions for additional data gathering? -bash-3.2$ zpool status zp1 pool: zp1 state: ONLINE status: The pool is formatted using an older on-disk format. The pool can still be used, but some features are unavailable. action: Upgrade the pool using ''zpool

Hi, I am looking for a softphone (sip or iax) that works in Solaris/SPARC with sunray100 terminals. I found iaxcomm but it doesn't work. Also I am trying sip-communicator but I have several errors from JMF/RTP. Does anyone have a softphone working over this platform? which one? I don't care if it is a commercial product, I can buy it if works fine. thanks in advance. Sebas --

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

Hi, I'm Computer Science undergraduate at The University of Cambridge, England. I'm currently writing a dissertation on using Vorbis for "Receiver Driven Layered Multicast". This uses a similar idea to bitrate peeling to create a multicast audio stream where the client can pick the bitrate by receiving a subset of the multicasted layers. I will be happy to release some

Hey all Ok, this is what i want to achive... Im actually developing an hybrid workflow controller, that uses JDF and JMF (see http://www.cip4.org/overview/overview.html#What_Is_JDF for more) - however, i have the situation where i need to send the JMF messages to my hardware output devices JMF portal over HTTP post and with a specified content-length header. Now, i know how to set headers

Hello, I have a small LAN network connected through an Asterisk Server. When I try to make a call between two of the user pc's on this network I get a "401 Unauthorized" error. Would anyone know how to remove the Asterisk Authorization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ;

hello, since while recoding asterisk is not sending an audio stream, the remote party times-out rtp and is hanging up on us. is it possible to send a blank audio stream while recording app is running? thanks, jrun

Hello everyone, since some weeks I experience strange problems on my gateways to the PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN What happens is, that after a while (uptime was a least two days) the gateway starts to not transmit audio to the PSTN on outgoing calls, but the caller can still hear the called