I'd like to broadcast an Asterisk agent's conversation to a ShoutCast server. As each queued caller gets through in turn to the agent, his current conversation with his current caller needs to be sent to a specific stream.
I can find lots of info on setting up Asterisk to receive a ShoutCast broadcast, and I can find some info on using Ices to send a MeetMe conference to ShoutCast. The latter is no good for me as I need the queueing system and as far as I can see there's no way to be in a queue and a conference at the same time. Any other information is eluding me.
Thanks.
[edit] - whilst I've said ShoutCast above, any broadcast server would do. Preferably one I can run locally.
[UPDATE] -
This question is no longer relevant to my particular problem. This functionality is no longer required for my project and therefore I don't need an answer anymore. However, it received 2 up votes so I can only assume that some other people would like an answer. Not sure of the etiquette in this case but should I leave it open so someone else can answer for others to reference?
You can use the asterisk ices command [1] or install a parallel freeswitch server, bridge both servers and use freeswitch's mod_shout [3].
[EDIT]
To capture the conversation use a conference room and setup a new call using a Local channel to the conference and to a dialplan context where you can run the ices command. [4]
More... you can use freeswitch and asterisk together to solve this. Make a call to freeswitch from asterisk Instead the ices command.
REFERENCES
[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Ices
[2] http://en.wikipedia.org/wiki/Icecast
[3] http://wiki.freeswitch.org/wiki/Mod_shout
[4] Join two conferences in asterisk
Related
Hi,
I'm busy developing a web interface for the asterisk PBX.
I'm looking for a way to initiate warm transfers via the web interface using the AMI.
I know that it's possible to initiate a warm transfer from the handset itself, but the requirement here is that it be done from the web interface.
I've done a fair amount of googling on the subject but I've not found anything thusfar.
Anybody know anything?
TIA.
To see all of the available manager commands, use the "manager show
commands" CLI command.
You can get more information about a manager command with the "manager
show command " CLI command in Asterisk.
https://wiki.asterisk.org/wiki/display/AST/AMI+Manager+Commands
You also can consult this page(note, that you still have check on your asterisk version like described above)
http://www.voip-info.org/wiki/view/Asterisk+manager+API
You are probably asking about Redirect command to some special context with dialplan support.
I am trying to collect some info about the ongoing call in asterisk but during hangup I want to log which peer initiated the process of hangup. I am new to asterisk so I have no idea if its possible or not. Please help me if it can be done. I have tried to use HANGUPCAUSE_KEYS but it does not provide much information.
Actually I want to know who has released the line first? If A and B are connected with bridge and B releases I want to record that B has released the channel and vice versa.
Thanks!
I can see 2 ways.
1) Set a Hangup Handler in your dialplan (extensions.conf). Maybe it is the easiest way, but limited. WIKI: Hangup Handler
2) Use AMI and try to monitor the channels. It can be complicated, but almost limitless and powerful. WIKI: Asterisk AMI
My description could be a little vague because I have not big experience in this field.
The problem is that my web service should do the following steps.
Another service send phone number in my web service
My web service takes that number and start calling into a particular queue in asterisk
After someone in this queue pickup call he\she should hear a recorded audio message
After that astersik should dial to the phone number from first step
Right now I can call to a local extension and then asterisk do the rest by calling to a client and connecting him with manager.
The first problem is that I don't know how to dial not to a local extension but to a queue in asterisk.
The second issue is how to play audio only when manager pick up call made from my web service.
Would be appreciate any help.
IF you use freepbx, you should put message in Call Confirm Announce
If you use custom dialplan, you should use M option for dial command and create macro which will play needed file.
For dial queue in freepbx you have use queue_num#from-internal. No way give any suggestion for custom dialplan
Note: doing system like that without understanding asterisk internal can result hi bills for international calls becuase of hackers.
I'm trying to complete software which does all call logic via AMI on it's own using Asterisk only as interface to VOIP, SIP/GSM. Almost everything works great, but...:
Here is my scenario:
- incoming call is forwarded to announcement and then to MOH forever
- my app decides which extensions to dial (7777) using AMI Action: Originate
- once somebody picks up on extension, his/her channel (SIP/306-xxxxx for example) is bridged with waiting call's channel using AMI Action: Bridge
Until this point everything is working fine, both connected parties can hear each other, recording on demand works. All is fine.
Now I'm trying to make assisted transfer to another extension (Atxfer) using AMI on one of the bridged channels. And it doesn't work. I got couple of ami events about DTMF's on a channel (audio is muted while they are played). Every DTMF digit couses quick Bridge:unlink and Bridge:link event on AMI.
I tried to change dtmfmode, upgrade from asterisk 1.8 to 11 (asterisk now) and it always was the same.
While having this problems with Atxfer blind transfer on those channels works (using AMI Action: Redirect).
full log shows nothing something like this:
[2013-11-11 20:24:57] DEBUG[9457]: features.c:3740 feature_interpret: Feature interpret: chan=SIP/306-00000017, peer=SIP/GTS-00000016, code=*2, sense=1, features=0, dynamic=apprecord#apprecord
I recommend you read some asterisk book for beginner like ORelly's "Asterisk the future of telephony".
In you case correct solution is use asterisk Dial command for first channel instead of second call creation.
It is not clear how you do transfer using AMI. If you want do it via ami(which is VERY bad way), you have do something like following
On transfer request(digit) unbridge channels. Better put it in AsyncAGI after that.
Collect digits where to transfer using Read command
Transfer to new destination
If fail bridge again
NOTE: You resulting application will be really buggy and not scalable. AMI interface is not designed to do such things and work very bad when you have alot of actions and channels running on same box. So you have test your app under hi concurrent load to ensure it work(or more likly not work).
I am trying to build an application where I am required to record and playback simultaneously. The application needs to go live on an asterisk telephony server. My problem is:
A user calls the asterisk server and starts to speak.
The voice packets being sent by the user are recorded in a wav file on the system.
A copy of the voice packets are sent as feedback simultaneously.
I have taken a look at ChanSpy, but it will not work if one is using Record.
My questions:
1. If a user calls an asterisk server, does that channel become a simplex or half-duplex channel?
2. Are there any commands etc. that allow us to do the above?
3. If not, does that mean I need to go into C programming for asterisk (agi-bin)?
P.S: Please let me know in case more information is needed.
Why not try MixMonitor? That allows you to record the call, and doesn't interfere with ChanSpy usage. ChanSpy IS the best way to do this, by the way.
You are wanting to feed to voice channel from the caller back to the same caller correct?
Have you tried the Echo command?