Our Asterisk PBX is set up with one extension "xxx" to dial an employee.
The logic in our diaplan will decide which channel to call for this extension, but there is no channel xxx.
This works fine for internal calling, but the queues are now all empty, since there is no channel "xxx" anymore.
I've tried creating an app_agent_pool and assigning agents with the xxx extensions, but whatever I do, the queues remain empty.
Is there any way to direct a call from a queue to an xxx extension that is not defined in sip.conf and is not a channel, and let the dialplan handle the rest?
Sure, you can use Local channel and dial to dialplan.
agent=> Local/xxx#mycontext/n
For normal operation you also should check hints&states
https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints
Related
I've been using a sim900 module to replicate many of the functions found in a basic cellphones for an embedded project. I've been successful with most functions with the exception of entering options during a voice call. I am actually looking for a generic solution (e.g. GSM 07.07 etc.) although the GSM/GPRS Module I'm using is the sim900.
Scenario: I initiate a call using ATD<number>; ,then automated voice asks me to dial "1" for an option. How do I send the "1"?
I've search high and low for an answer. I've been through the AT command manual over and over again. Please help.
Very good start in using the official GSM specification, although I want to note that 07.07 has been superseded by 27.007 a very long time ago, and you should use that document (or 27.005 if relevant).
I initially expected there to be two ways of achieving this, dial string modifiers or DTMF command, but I looking up the dial string in the 27.007 specification I do not find the p (pause) modifier1 I was expecting, and nearest thing, the W (wait) modifier is ignored and only included for compatibility.
Sending 0 through 9, A through D, * and # during a call is done using DTMF, although in a GSM network this is signalled separately out of band rather than sending in-band analogue tones. There is a specific command for sending DTMF tones called AT+VTS (with a horrible syntax). So that command is the answer to you question. Invoke it either from online command mode or from another serial connection.
1 The reason I was expecting a p modifier to exist is that I am able to enter one in phone book entries, e.g. `"12345678p123" which will dial 12345678, wait for the connection to be established and then send 123 as DTMF tones. But this is then obviously something (only) the user interface voice call handler manages and not the AT command handler.
Help me track the status of a specific port: "LISTENING", "CLOSE_WAIT", "ESTABLISHED".
I have an analog solution with the netstat command:
local command = 'netstat -anp tcp | find ":1926 " '
local h = io.popen(command,"rb")
local result = h:read("*a")
h:close()
print(result)
if result:find("ESTABLISHED") then
print("Ok")
end
But I need to do the same with the Lua socket library.
Is it possible?
Like #Peter said, netstat uses the proc file system to gather network information, particularly port bindings. LuaSockets has it's own library to retrieve connection information. For example,
Listening
you can use master:listen(backlog) which specifies the socket is willing to receive connections, transforming the object into a server object. Server objects support the accept, getsockname, setoption, settimeout, and close methods. The parameter backlog specifies the number of client connections that can be queued waiting for service. If the queue is full and another client attempts connection, the connection is refused. In case of success, the method returns 1. In case of error, the method returns nil followed by an error message.
The following methods will return a string with the local IP address and a number with the port. In case of error, the method returns nil.
master:getsockname()
client:getsockname()
server:getsockname()
There also exists this method:
client:getpeername() That will return a string with the IP address of the peer, followed by the port number that peer is using for the connection. In case of error, the method returns nil.
For "CLOSE_WAIT", "ESTABLISHED", or other connection information you want to retrieve, please read the Official Documentation. It has everything you need with concise explanations of methods.
You can't query the status of a socket owned by another process using the sockets API, which is what LuaSocket uses under the covers.
In order to access information about another process, you need to query the OS instead. Assuming you are on Linux, this usually means looking at the proc filesystem.
I'm not hugely familiar with Lua, but a quick Google gives me this project: https://github.com/Wiladams/lj2procfs. I think this is probably what you need, assuming they have written a decoder for the relevant /proc/net files you need.
As for which file? If it's just the status, I think you want the tcp file as covered in http://www.onlamp.com/pub/a/linux/2000/11/16/LinuxAdmin.html
In all Diameter implementations I saw, the messages originating from the server is always sent towards the DNS resolved IP address of whats in the Destination-Host AVP. But, in commercial servers, we see an option to configure a DRA or a DEA which takes in all the messages and routes them.
Thus, when it comes to the mobicents diameter stack, this approach is sometimes hard to do. I can anyway re-configure the hosts file so that the message ends up in a DRA/DEA, yet, its a pain. I see no option to send these messages to a central diameter agent which will take care of all the dirty work for me.
The next issue is, if I plan to create such a DRA/DEA, the stack does not accept messages to a different host. Where, the message's Destination-Host parameter might contain a different hostname than ours. (which would be the ultimate destination it needs to go)
Is there a hack to achieve this without meddling with the internals of the jdiameter code and RA code?
If you change jdiameter's config to something like this:
<Network>
<Peers>
<Peer name="aaa://127.0.0.1:21812" attempt_connect="false" rating="1" />
<Peer name="aaa://CUSTOM_HOST:4545" attempt_connect="false" rating="1" />
</Peers>
<Realms>
<Realm name="custom.realm" peers="CUSTOM_HOST" local_action="LOCAL" dynamic="false" exp_time="1">
<ApplicationID>
...
</ApplicationID>
</Realm>
</Realms>
</Network>
In your sbb, then you'll need to create a client session providing your custom realm using this method:
DiameterCCAResourceAdaptor.CreditControlProviderImpl.createClientSession(DiameterIdentity destinationHost, DiameterIdentity destinationRealm)
Example:
ccaRaSbb.createClientSession(null, "custom.realm")
where ccaRaSbb is a CreditControlProvider instance (resource adaptor interface)
finally, when creating your CCR, the method CreditControlClientSession.createCreditControlRequest() will use the session' realm to find an available peer previously configured.
Let me know if this makes sense to you
Posting the method I used to solve this problem.
As it turns out its not possible out of the box to send a diameter message towards a peer which is not configured in the stack's jdiameter-config.xml file.
For me, the option to alter the stack in this case was also not feasible. So I devised a workaround for the problem by co-operating with the DRA we have. (most DRA's should be able to handle this method)
I added two custom AVPs to the outgoing request, namely Ultimate-Destination-Host and Ultimate-Destination-Realm.
In the DRA, I asked the admin to delete my Destination-Host and Destination-Realm AVPs and replace them with the ones created in step 1.
Now, whenever I send a packet destined to other diameter peers outside the configured peer, I target them towards the DRA and set these 'Ultimate' destination AVPs.
Ours is an Oracle DSR which is capable of doing this AVP manipulation. Most commercial ones should be able to handle it. Hope someone who wanted an answer for this question found this useful.
I need to strip the rport parameter from the Via field of SIP messages being generated by Yate (for compatibility with a broken peer). Can I use the scripting capabilities of Yate to do this? How do I intercept and modify outgoing SIP messages?
I discovered, by reading the code, that Yate already supports this feature. In accfile.conf, in the section defining the sip server to register to, place the following line:
xsip_flags=1
This prevents the rport parameter from being placed in SIP messages. This may break routing if you are behind a NAT, so beware.
An example config would therefore be:
[sip_service]
enabled=yes
protocol=sip
description=sip_service
username=user
domain=somewhere.com
authname=auth
password=secret
server=somewhere.com
xsip_flags=1
I'm trying to initiate calls using the ARI API, the process I followed was
POST /ari/channels to create channel 1 to the local extension
POST /ari/bridges to create a bridge
POST /ari/bridges/{bridge-id}/addChannel with channel 1
POST /ari/channels to create channel 2
POST /ari/bridges/{bridge-id}/addChannel with channel 2
I have a websocket connection waiting for the correct events (like channel status up) before calling the POSTs.
The problem I'm having is this works using internal extension numbers for channel 1 and channel 2, but when using an external number for channel 2 I get the error - "Allocation failed".
Any ideas where I'm going wrong...
Im creating my calls via ARI like this:
http://{ARIUser}:{ARIPass}#localhost:8088/ari/channels?endpoint=SIP/{exten to call from}&extension={number/exten to call}&context=from-internal&priority=1&callerId={callerID}
When you create channel with ARI to address "/ari/channels" you MUST define "extension" or "application" parameters (read more https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Channels+REST+API#Asterisk12ChannelsRESTAPI-originate).
Digitum make very unusable Python ari module and you may play with my module (now not implemented events) on https://github.com/vit1251/ari2.