asterisk

Closed. This question is off-topic. It is not currently accepting answers. Learn more . Want to improve this question? Update the question so it's on-topic for Stack Overflow. I want to configure Kamailio server so that traffic will be forwarded to other four asterisk servers equally. It is working fine with a single asterisk box but I am unable to forward a call to another asterisk box. Here is the kamailio.cfg that I am using. #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_ASTERISK # *** Value defines - IDs used later in config #

I need a help with executing asterisk cli commands using C#. I'm able to open terminal window and start asterisk (see the code below), but dont know how to for example execute "sip show peers" command in CLI. Is there any possible way to do it? using System; using System.Diagnostics; namespace runGnomeTerminal { class MainClass { public static void ExecuteCommand(string command) { Process proc = new System.Diagnostics.Process (); proc.StartInfo.FileName = "/bin/bash"; proc.StartInfo.Arguments = "-c \" " + command + " \""; proc.StartInfo.UseShellExecute = false; proc.StartInfo

I am doing something about ConfBridge these days. I used "confbridge kick all" to end the conferences when admin user exit, but the others in the conference will hear "you have been kick out from this conference.". it is not a good way to end a conference. Is there any configure or command which can set that conference will end up when the admin user exit the conference? Just set endmarked=yes on the user profile you're using (found in confbridge.conf). The docs say: ;end_marked=yes ; This option will kick every user with this option set in their ; user profile after the last Marked user

Salvete! How can I dial a number and have Asterisk originate a call from extension sipX to sipY? Both sipX and sipY appear in extensions.conf of my dialplan. The trick is that I want to dial 337 on my phone, and then my phone goes out of the picture, then sipX calls sipY. Say I want to be able to push 337 on the phone, and have a sound played over the speakerphone of another phone, say, as an alarm. Another way to consider it is, how can I do this: Push 337 on my phone complete my call, that is, hang up the original caller run a macro even though there is no call in queue call sipY play a

My objectives: Client(SIP phone, i use 3CX phone) dial to asterisk server, asterisk then connect with external relational database(not located in the same place with asterisk server), and if database response something, asterisk server play a voice file(predefined .gsm file) to response client. What i already have: I have installed AsteriskNow in VirtualBox as a asterisk server, the client is using softphone to connect with Asterisk server in SIP channel. When the Client dial to asterisk server, the server can execute the dialplan. My question: If i want asterisk server establish connection to

So I have a VOIP system set up through a FreePBX server. I want to have it so that when a new call is picked up by FreePBX, asterisks will send the caller ID and the call ID to a php script, which will then use that information to gather ticket information for the account related to that caller ID. It will then update a database with the found information. When a user answers the phone, I then want to send the user's extension and the call ID to another php script and update the database with the new information. I have looked into PHPARI, but the documentation is lacking for me. I just need

I'm trying to use sox to convert asterisk voicemails to raw audio. The encoding information of the original wav is PCM S16 LE, so I thought I would just be able to do sox msg0000.wav msg0001.raw but the raw file from that is garbled and, according to VLC, 4 minutes long, compared to a 6 second source file. I'm not sure where I'm going wrong, anybody how to convert a wav to raw? Preferably using sox, but any commandline solution will do fine. Are you sure that your problem is with sox and not VLC? I've also had problems getting VLC to play raw audio. There are command-line options but I didn't