Everyone,We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here:http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asteris..
Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E cant hear, is there a way to suspend D and E for a while (whilst t..
Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non provisio..
HiRe raising this issue as its still affecting me.Where is the asterisk server getting port 0 from? We use ARA and port 0 is neither in the full contact not in the port field of the sip table. Nor is port 0 in the realtime cache for any peer register..
Greetings to all. I am not sure of this is a user question or a business so apologies in advance if it should be asked in the business list.A client of mine has a UK branch that is served by a provider that uses the Broadsoft solution. I want to cre..
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I am having trouble compiling dahdi-linux-complete-2.9.0+188.8.131.52 on a Raspbien 3.10.25+ kernel. I get the following error – /usr/src/dahdi-linux-complete-2.9.0+184.108.40.206/linux/drivers/dahdi/dahdi-base.c:570:2: error: implicit declaration of function kzal..
Im running a test system with Ast 11.7 and DAHDI 2.9.0. I loaded a TE205Pcard. Then I enter asterisk console, and once I do a pri show channels, the console no longer works correctly. Theres no output for any command after that. If I type pri show ..
This is evenily.My one Asterisk server and event listen App was working well for several month. Yesterday My event listen app stop work suddenly. I telnet to AMI via localhost 5038 to check the events, and I find Asterisk does not push the events un..
Hey all,Ive been fighting with this all morning, and I feel like this should be a relatively simple task, but I just cant get it to work.I currently have a very basic asterisk v11.6 setup with a single extension (a Bria softphone) and a single sip tr..