asterisk

Only One Side Recorded With Asterisk Manager API

I am using Asterisk-Java to send Manager API calls to monitor phone calls. For some reason, when the file is finished, I only end up with one side of the conversation recorded. Does anyone have any idea why this might be happening? Here is my call: Response response = connection.sendEventGeneratingAction(new MonitorAction(theChannel,...

Type Declaration - Pointer Asterisk Position

Hello, in C++, the following means "allocate memory for an int pointer": int* number; So, the asterisk is part of the variable type; without it, that would mean something else (that's why I usually don't separate the asterisk from the variable type). Then what is the reason the asterisk is considered something else, instead of being p...

Asterisk TDM Out Channel Not Recording

I am trying to use the monitor command to record a TDM extension, but only the in chnnnel is being recorded. The out channel is 44 bytes and obviously no audio within. However, when monitoring a SIP or IAX phone, no problems exist. Is there some configuration I'm missing for distinguishing between TDM and SIP/IAX for recording? Thanks i...

How to receive the text which was sent by SendText

In asterisk I have sent a text message using SendText as follows I have two registered users in sip.conf file. sip.conf details [thillai] username=thillai secret=thillai host=dynamic type=friend allow=all context=test [selvan] username=selvan secret=selvan allow=all host=dy...

Asterisk + FreePBX + GoTalk. Inbound route not working.

I'm running asterisk 1.6.2.6 and freepbx-2.7.0 My trunk is configured as follows: Outgoing Settings Trunk name: GoTalk Peer Details: host=sip.gotalk.com username=09xxxxxx secret=YNxxxxxx type=peer fromuser=09xxxxxx fromdomain=sip.gotalk.com canreinvite=no insecure=very Incoming Settings User Context: 09xxxxx User Details: use...

How can I ring a dial group and land everybody that answers into a conference room?

I'm using FreePBX from a Trixbox install to manage an Asterisk server. I added a dial group with ringall strategy, but as soon as one person answers, the other extensions in the group are dropped. I'd like to keep ringing these extensions so that everybody that picks up the call lands in a conference with the caller. It would be accepta...

BackgroundDetect() application in Asterisk

Hi All, I'm learning Asterisk. In that I started to learn about BackgroundDetect() application. There are three options are there. BackgroundDetect(filename[|sil][|min|max]]) sil - If we specified 1000 in sil option,it'll wait 1 second after we say something to phone. I worked sil option,it was working fine. But I didn't understand...

asterisk : add application

Hi all, I want to know the way to add new asterisk applications and modules.For example I don't have the SetGlobalVar application in my asterisk machine.I want to add that.Is there any way. Thanks in advance . ...

asterisk : Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

Hi all, I am learning asterisk. After I have installed asterisk I have tried to connect with it using asterisk -rvvvvc. But it gave me an following error message. Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) How can I solve this issue.Please help me. Thanks in advance. ...

SpeechBackground

Hai everyone, I have used the SpeechBackground application in asterisk. I used the version 1.6.0.6. I have a entry like, ;;SpeechCreate exten => s,1,SpeechCreate() exten => s,2,SpeechActivateGrammar(yesno) exten => s,3,SpeechStart() exten => s,4,SpeechBackground(demo-instruct) exten => s,5,SpeechDeactivateGrammar(yesno) I don't kno...

Asterisk : Meetme application is not found

Hi all, I am learning asterisk. I learned about the application meetme which is used to make a conference call. I am using asterisk 1.6. In that the application meetme is not found.What shall I need to do to work with meetme application. Please Help me. Thanks in advance. ...

Looking for an explanation of Asterisk's cdr log fields

Asterisk has the following fields CREATE TABLE `cdr` ( `calldate` datetime NOT NULL default '0000-00-00 00:00:00', `clid` varchar(80) NOT NULL default '', `src` varchar(80) NOT NULL default '', `dst` varchar(80) NOT NULL default '', `dcontext` varchar(80) NOT NULL default '', `channel` varchar(80) NOT NULL default '', `...

What is mean by Action => 'ping' using send_action in Asterisk Manager Interface script?

Hi All, I have started to read about Asterisk::AMI module. In that module if we want to send the action to the AMI server,we need to use the Action with action name using send_action method. In that module they mentioned about Action => 'Ping' within send_action method. Here what is the use of Action => 'Ping'.Can anyone explain me abou...

Asterisk: Originate API - Which card to use to detect busy/ringing/answer event for FXO

I want to use Originate API of Asterisk to place an outbound call on a FXO channel, for testing purpose I am using X100P card and, as expected, card is not able to detect if the number is busy/ringing or when it is answered. I want to know which card should I use so that I can get such basic events ... I am not really interested in deta...

How to handle DTMF on outcall using call files in asterisk?

Hi, I'm making an outdial using call files in asterisk and application needs some DTMF input but DTMF not working for all mobile phones, It is not accepting digits from Nokia - 1100 and nokia 6030 where DTMF works if i make an incoming callfrom the same phone but on out dial using DTMF log i can see following messsage on asterisk CLI: ...

how can I add users/numbers to asterisk from php?

How can I go about adding users/number, changing things from PHP(or python) on an asterisk server? ps. also are there any better ways to get the current asterisk settings, users, numbers other than scraping the config files? ...

Adobe stratus VS Asterisk

What is the best way to create a simple application that make PC-to-PC call ? Why Stratus is better ? Why Asterisk is better ? Remember that stratus use a peer-to-peer protocol. ...

Originate call using adhearsion

i have searched google, tried to read adhearsion docs and make sense of the examples. i understand parts of it but am not able to put it together. i can edit extensions.conf and do agi(agi://localhost) and handle incoming calls in the dialplan.rb. i have written an ivr that way which is in use in the real world. i have edited manager...

Asterisk Manager API SIPPeers - Permission Denied

I'm wanting to use the asterisk manager api to show the status of all my SIP lines in a PHP web interface. I thought I'd start simple and use telnet to see it working. So I created a user in /etc/asterisk/manager.conf [portal] secret = password read = all,system,call,log,verbose,command,agent,user Then telnet to localhost on port 503...

How to send REGISTER request periodically from my SIP client to Asterisk server using Asterisk Manager Interface

Hi, I am using Asterisk 1.4 server and I have created a desktop client using the Asterisk.NET Library. I am able to log into the AMI (as a manager) using Asterisk.Net, but I cannot find a way to send the REGISTER command using the AMI, to share my client's location information with the server. I want to know an AMI or a CLI command tha...