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).
Related
I have a working system that controls a Cisco CUCM IP-PBX to set up and tear down a call between two parties A and B; it makes use of Java's JTAPI to:
make A call B
make B answer (pick up)
(wait for a few seconds)
make either A or B drop the call
Now I want to do the same with an Asterisk PBX (version 13.17.0). From what I have gathered, Asterisks JTAPI implementation doesn't work for recent Asterisks; the latest version that I can find, dates from 2006 and it seems to try and send AMI commands to which Asterisk replies something along the lines of "no such command" - I'm guessing the AMI syntax has changed over the years.
What I'm using now is a library called asterisk-java and more specifically, its AMI support. I'm sending an OriginateAction and the net effect is that:
Asterisk calls A
I have to pick up A's phone manually
A calls B
I have to pick up B's phone manually
finally a call is established between A and B
So instead of a fully automated process like in the Cisco case, I have to do manual work and I can't seem to find an API to just set up a call between A and B and having it drop after a couple of seconds.
Questions: is the AMI the correct API for my use case or is there a better alternative? And what API calls do I have to make to establish and drop a call?
Thanks,
Jan
Does this answered question help with establishing a call without picking up:
Originate a call with Asterisk - without the originating extension ringing
To have a channel drop a call, using AMI, you can use 'BridgeKick' to boot a channel, or 'BridgeDestroy' to cause each channel to 'BridgeLeave':
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_BridgeKick
NET experts,
I have a scenario where a 4-port PSTN card is installed in a server and I have installed Freebpx on that server as per suggestion by someone. When a call comes on any of the PSTN line, it is forwarded to one of the operators on his hard phone.
Each operator is also having a computer screen at his table, powered by an individual CPU. This runs our CRM software to be handled by operator. When a call arrives to an operator hard phone, say operator 2, we want that the Caller number should also be displayed in the CRM software. Based on this caller number, operator can enter some information related to the Caller and save it in database via our CRM software. Also, when operator disconnect the call, we should receive call stop time for statistics later on.
Thus, we need caller number and call start time when a call is picked by an operator on his hard phone. and then we need call end time when a call is finished.
Can someone help us how we can achieve this? Do we have to capture the SIP packets and parse them or their is some other way to do so? Our CRM database is totally separate from the Freepbx and resides on another server.
If you want to get these events realtime, you should look at AMI (Asterisk Manager Interface - port 5038 TCP by default) and it's configuration manager.conf (note: FreePBX uses the, see manager_custom.conf).
If you want the archived version, you should set up a database server, and point the CDR (Call Detail Records) module to it. PostgreSQL or MySQL/MariaDB works just fine. Asterisk will simply ignore additional fields in the CDR as long as they can be NULL or has a DEFAULT value. This can be used to store custom data.
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 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?
I'm trying to setup an IVR or to be more specific an automated attendant using Asterisk. I don't want anything fancy besides the simple automated menu system, and instead of dialing an extension (for now) just ring the same phone line (POTS) if pressed 1 or play a recording if pressed 2. The server running Asterisk will be in a remote location.
My questions are:
Is this setup doable?
Do I need to sign up with a IAX VOIP
provider? (no VOIP calls will be
made/received) Only normal POTS calls will be made. (possible in other country)
What hardware do I
need? Remote server side and client-wise
(POTS phone line)
Any comments will be appreciate it.
Thanks
This is definitely doable, however, it seems that your specifications need to be modified a bit. Here are some things to consider:
What type of PSTN connectivity will your remote Asterisk server have? (SIP / POTS / T1 / PRI / etc.)
If your remote Asterisk server is going to be using a physical medium, do you have the connectors and hardware in place? EG: If you are using a T1 line, do you have a channel bank or T1 card?
Are you comfortable with Asterisk dialplan / AGI / AMI, or are you going to use an Asterisk distribution like trixbox, AsteriskNOW, Elastix, etc?
Will your client location (with the POTS line you wish to ring) have a PBX, or will it just be a typical POTS line hooked up to an analog handset?
My recommendation to you:
Get a cheap server (any 1U with a dual core processor and 512MB of RAM will do), and put it at your remote location.
Load Asterisk 1.6+ onto your server. I recommend 1.6+ as it can use the dahdi_dummy driver as a reliable software timing source (it will ensure that your audio quality is not choppy and broken).
Get a SIP account with a reliable SIP provider. My personal favorites are: flowroute and voipms.
Set up your new SIP account in Asterisk, and purchase a DID (phone number). This phone number will be your business phone number, the one that you give out to clients and put on business cards.
Configure your Asterisk dialplan to receive calls from your SIP account to your IVR menu.
Your IVR menu logic should be something like:
a. Play the IVR menu.
b. Wait for a keypress.
c. If the user dialed '1', then make an outgoing SIP call to the POTS line phone number you want to reach. If the user dialed '2', then playback the recorded message.
Now, if you are looking to save money, and have the most cost-effective setup for your remote IVR, I would recommend throwing up a second Asterisk server on site at your client location (where the POTS line comes in), and throw away the pots line and just setup an IAX2 trunk between your client location and your hosted server location. This way, when calls come in to your remote Asterisk server via your SIP provider, you can route the calls (when the user hits option 1) over your IAX2 trunk, directly to the client location for free!
Depending on your skill level, and comfort with Asterisk, this could be either a really fun learning experience or a confusing nightmare. If you would like to learn more about telephony and Asterisk, especially if you are going to use it for your business, you may want to use a simple (free) Asterisk distribution like: trixbox CE, Elastix, or AsteriskNOW.
You could code up a simple IVR in less time than it takes to install Asterisk if you use a hosted service like Twilio or Tropo. I'm partial to Twilio myself.
Edit: here's an example of a simple phone menu.
Quick answers to your simple questions:
this setup is doable.
no, IAX is not neccessary
some hardware setups are listed in wiki
I think you should start with wiki and free Asterisk book