I'm adjusting simple application that among other things should be able to call another party using Asterisk AMI Originate command.
I'm stuck and I believe that my originate command is wrong.
Where/how can I see log of Originate commands that Asterisk creates when I use regular phone so I can compare it to my hand crafted one?
Use a network sniffer, such as tcpdump or wireshark, and capture the packets that come and go to/from asterisk. By default, it uses 5038/tcp. Check your manager.conf file, and look for the bindaddr and port options to be sure you capture the right traffic.
If you are using ssl (sslenable=yes), then you will have to configure wireshark with your ssl keys, so it can decrypt the traffic or just use normal tcp (without ssl) for debugging and then switch to ssl.
You should see the Action: Originate coming in to asterisk, and the asterisk response and the associated events. Look for the ActionID parameter of the action so you can trace which responses and events correspond to each issued action.
Take into account that an async originate (async: true) will return a response as soon as the action is received by asterisk, but it will then send asynchronous events to inform the call status (once finished). On the other hand, when using async: false, the call will be placed and the response will have the status.
A few more resources on the originate action:
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
https://wiki.asterisk.org/wiki/display/AST/ManagerAction_Originate
Another question related to the async parameter:
Asterisk originate response says successfully queued but nothing more
Hope it helps!
EDIT: Asterisk does not create the originate command, but will dial a target (a channel) based on an incoming originate action, or call file, so your application (the ami client) will issue an originate action and then asterisk will react to it by doing the call. If your call is originating from a phone, it's more probable that the call is being originated by a dial() command in your dialplan.
Related
I want to setup and IVR Menu i mean if a user calls to a particular GSM Number then the number should be redirected to Asterisk Server and the user needs to Get IVR Menu
I am using VoiceBlue Next firmware version 1.31.1.34.1 inserted working SIM Card
If i make a call to that particular number i am able to accept call,reject call and other options from VoiceBlueNext Web Interface.
I have made a SIP account in pjsip.conf file and created and extension as 100 in extensions.conf but unable to transfer the call to Asterisk Server
In asterisk server are there any other files to be changed or any settings in VoiceBlue Next
There are not many details to understand your scenario, I have not used VoiceBlue but on Asterisk if you want to receive calls, from your VoiceBlue or any other provider. You have to do two things, one you have to register this peer to allow receive calls, or you can also set allowguest=yes(but very dangerous anyone can send you calls) or add peers at end of pjsip.conf file as little secure way.
Next, you need to add dialplan, suppose if you get any number _X will be any number, now you can put Dial your extension to receive any number from the provider.
As for sip client to call out you have to register peer and both must be in the same context.
Sending outgoing calls, now if you call any number beginning 6 and 7 they will be forwarded to VoiceBlue
exten=>_6XXXXXXXX,1,Dial(SIP/${EXTEN:0}#10.0.0.20,,r)
exten=>_7XXXXXXXX,1,Dial(SIP/${EXTEN:0}#10.0.0.20,,r)
for incoming please add following in your pjsip.conf
[VoiceBlueNext]
type=peer
host=10.0.0.20
username=voiceblue
secret=password
fromdomain=10.0.0.20
and in same file on top put following general section
[general]
port = 5060
bindaddr = 0.0.0.0
allowgues=no
context = sip
disallow=all
allow=ulaw
Notice I allowguest = no , so you must provide peer VoiceBlue peer information to receive calls, but if you want to test, make it yes and you will get calls without any security.
I am using Kamailio 4.4 as the proxy with my Asterisk server. I am trying to develop a scenario where an extension gets registered on asterisk via Kamailio when it receives a push notification. This push notification is sent to the sip extension when a call towards this extension reaches to the Kamailio.
For example, suppose there is two SIP extension( extension 1 and extension 2) registered on Asterisk via Kamailio. When a call from extension 1 reaches the asterisk, it forwards the INVITE request towards extension 2 via Kamailio.Kamailio will try to forward it to extension 2. suppose the extension 2 is not able to receive the INVITE request from Kamailio. When extension two receive a push notification, it will register on asterisk.
So I need to get the call on extension 2 through the new registration.
We are trying to simulate registration of extension to the asterisk when receiving the push notification.
First, we registered extension 2 and disconnected the network. Then we tried to register the same extension when a call from extension 1 reaches to Kamailio. This is a simulation of push-based registration since an extension that receives the push will attempt to register when an incoming call is received.
When asterisk sends INVITE request to Kamailio, it immediately responded with 100 trying provisional response. This 100 response by Kamailio towards asterisk prevents asterisk from re-transmitting the INVITE.
Then Kamailio tried to send and retransmit the packet to extension 2, which does not have network access. This extension 2 was on port number 24071. Even after successful registration(in port 59995) of the extension 2, Kamailio continued to transmit the packets to the old port.
After that, we have configured Kamailio in a way that it won't send an immediate provisional reply(100 trying ) for INVITE request.
Here Kamailio is not immediately sending 100 trying message to Asterisk. This forces Asterisk to re-transmit. Asterisk was found to retransmit the same packets. However, even after the successful registration of extension 2, asterisk continued to send the old invite to Kamailio not the new one to the latest port.
This is the problem for me since push relies on the INVITE reaching the phone at the correct port number.
So, is there other good approaches to solve this issue?
One thing I would like to try is modifying the pending INVITE request towards old registered port with the new port details when new registration reaches to Kamailio. Can I get the ongoing requests from Kamailio?
Please suggest a viable solution.
Almost any kamailio config availible do similar thing.
You have save into location and consult it when do call.
However if you need really scalable platform you SHOULD NOT forward register requests to asterisk at all.
If kamailio send invite to wrong port, likly that mean you have TWO records in location.
I have a java stasis application on Asterisk 14 using ari4java. It mostly works great. I am now trying to receive an external call and relay it back out. I do following
Incoming call enters Stasis
Create bridge
Add first call(channel) to bridge
Create channel
Add second channel to bridge
Dial( secondChID, "Local/2601", 30)
No matter what I try, the second outbound call gets the callerID of the first inbound call. That is actually OK for many calls, but in this case I want to set another callerId.
Before Dial() I have tried to setChannelVar(CALLERID(num)) and this value I can see in all events coming from Asterisk. But once the SIP call is placed, no sign of my callerID.
I doubt it is the ari4java doing anything wrong as I see the callerID in all the "dial" events. I thought I could force a callerID in sip.conf, but unable to do that too.
I want to send a fax with my application via Asterisk. I need to execute Originate command to send a fax and use Elastix virtual fax.
My Elastix virtual fax is defined IAX and 999 extension number.
Commandline:
asterisk -rx "channel originate Local/1234567890#from-internal extension 999#from-internal"
pro-sip*CLI> channel originate (tab pressed)
There are two ways to use this
command. A call can be originated between a channel and a specific
application, or between a channel and an extension in the dialplan.
This is similar to call files or the manager originate action. Calls
originated with this command are given a timeout of 30 seconds.
Usage1: channel originate application [appdata]
This will originate a call between the specified channel tech/data and
the given application. Arguments to the application are optional. If
the given arguments to the application include spaces, all of the
arguments to the application need to be placed in quotation marks.
Usage2: channel originate extension [exten#][context]
This will originate a call between the specified channel tech/data and
the given extension. If no context is specified, the 'default' context
will be used. If no extension is given, the 's' extension will be
used.
Sure this example not check anything like channel not availible or busy etc.
You can do same using AMI action originate
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
Correct application should also check dialling state, redial etc.
I have an orchestration that takes a message. The target namespace is "http://microsoft.com/HealthCare/HL7/2X" and the root element is "ORU_R01_23_GLO_DEF"
In the orchestration, I map the message to an intermediate message type in a construct shape. The target namespace is "http://mycompany.com/myapplication" and the root element is "MyMessage". The "MyMessage" message is then further mapped and then sent to a web service using a logical send port in the orchestration. A WCF send port is then bound to the orchestration and everything works fine. Everything works as expected.
Without altering the orchestration, I want to create a send port that subscribes to the intermediate "MyMessage" message and writes it to a file. To do this, I have created a send port with a filter of BTS.MessageType = http://mycompany.com/myapplication#MyMessage.
Even though messages are flowing through the orchestration, my send port isn't picking up the message. Is this the incorrect filter to use?
Are you trying to subscribe to the 'MyMessage' message, or the same message that is sent to the logical Send Port bound to the physical WCF Send Port?
You have stated that:
The "MyMessage" message is then further mapped and then sent to a web service using a logical send port in the orchestration. A WCF send port is then bound to the orchestration and everything works fine.
Message not Published to MsgBox
From what you have described, I would suggest that you do not have a Send Shape/Logical Send Port combination in your orchestration for the 'MyMessage' message, which is why you can't manually subscribe to this message type in a Send Port filter. The fact that you have not mentioned a 'Failed Routing Report' message further suggests that this is the case - this message type is generated when no subscriotion can be found for a message that is to be published to the MsgBox.
Capture a Message's 'MessageType'
If however you need to capture a copy of the message your are sending over the WCF Send Port, you will need to determine its MessageType and use that in your second Send Port subscription that writes the message out to file.
If you are unsure as to what MessageType to use, there is a simple trick to determine this information:
Stop (not Unenlist) the WCF Send Port
Send a message through your orchestration as normal - the message will be marked as 'Suspended Resumable' in the BizTalk Admin Console on the WCF Send Port.
Open the message in the BizTalk Admin Console and view its 'Message Context'; in the Message Context you will see its 'MessageType' property which you can then use to understand which subscription filter to use.
Start the WCF Send Port to flush the message.
Alternatively, if you don't want to change your orchestration, you could try archiving your message as it passes through the Send Pipeline in the (original) WCF Send Port - either write your own archiving component or use an existing commercial component. By using an archiving component in this manner, you will save yourself the expense of an extra subscription and the associated Send Port maintenance.
Update:
It sounds very much like the OP is not sending the intermediate message to the Message Box from their Orchestration (see comments). Message subscription will only work when a message is published to the Message Box - in this case, the message in question ('Message B') is an intermediate message that only lives within the context & lifetime of the orchestration. The OP needs to Send the message to a Direct Bound port within the Orchestration to allow the message to be subscribed to via a Send Port.
Verify the pipelines of the Send Port. Should by XML, not Passthrougth.