In my Asterisk 1.8 setup, MixMonitor() in the dial plan successfully records calls without issue.
However for queue calls, while a recording file is generated and the log shows that MixMonitor() is recording, however the files are tiny and contain no audio.
This is my queue:
member => SIP/extension123
monitor-type=MixMonitor
monitor-format=wav49
I have also tried changing the monitor type to "Monitor" and the format to "wav", with the same result. I have also tried adding local channels as queue members and recording from the local channel in the dial plan instead of using the queue's recording and I get the same result (even though when dialing the local channel extension directly the recording works fine).
Any help would be appreciated.
Update: I am trying the same thing now on Asterisk 11.7 and experiencing the same issue
Figured it out: calls were being passed to the queue via a local channel. Apparently you need to add "/n" to the local channel call to keep it from being "optimized away".
For example:
Dial(local/7001#Queues/n)
Source: http://svnview.digium.com/svn/asterisk/branches/1.6.1/apps/app_mixmonitor.c?view=markup&pathrev=197897
Related
I have a FreePBX 13 server set up with a SIP Trunk connection, however for some reason we are not getting the ring back tone for calls going out of the trunk connection.
I was able to implement a work around for this by placing the "Tr" options under "Asterisk Trunk Dial Options" to force Asterisk to produce the ring back tone for outbound calls.
However this only works while manually dialing from a soft-phone / VoIP Phone, when I try to launch a call via the Asterisk AMI "Originate" command we are not getting the ring back tone, even when the "r" option is set on the trunk. This is how the command I'm sending looks:
Action: Originate,
Channel: SIP/{extension},
Context: from-internal,
Exten: {phoneNumber},
Priority: 1,
Callerid: {callerId},
Timeout: 30000,
Async: yes
Any idea about what can I do to force the ring back tone?
This could be due to inband progress.
Try to add the following to your sip.conf
under [general] section,
prematuremedia=no
progressinband=yes
if you are using freepbx, then navigate to your trunk settings in the user interface and add the above 2 configs under the peer settings.
restart asterisk and try after that.
Launch Originate via Local/ channel call, if still not work, launch via custom context which answer first.
This is not an uncommon problem. Your ITSP is being lazy and either not providing or not forwarding RFC-complaince SIP 180/183 responses.
My usual solution is to put the local caller into an MOH state where the MOH is a ring tone, and then when the other end picks up, bridge the two calls.
This is kind of jarring, however, if the remote end is busy or is congested, since you go from "fake ring" to "real error tone". The other option is to reverse your dialing process ... call the remote end and then hook the local end, so that the local end doesn't need to hear ringing.
Of course, you can also reach out to your ITSP and tell them you need them to be RFC SIP 180/183 complaint.
I am a real beginner in asterisk, so please tolerate my question :)
I tried to configure asterisk for realtime and it is working fine for local sip calls. Now, I am trying to make the following test with dahdi calls:
I connected an analog phone to an FXS channel of my Digium card and tried to call this phone (exten 124) from a sip softphone (X-lite).
I get the following error:
-- Executing [124#from-sip:1] Dial("SIP/2000-00000004", "SIP/124")
[May 31 10:24:22] WARNING[5457]: chan_sip.c:5667 create_addr: Purely numeric hostname (124), and not a peer--rejecting
my extensions.conf:
[from-sip]
switch =>Realtime
[from-pstn]
exten => 124,1,Dial(DAHDI/3)
It seems that the dial is done using from-sip context not from-pstn context as required.
Anyone to advise or correct my understanding?
Thanks million
Zak
In Asterisk realtime and not realtime you can configure where to send calls from particular extension, this should be configured in "context"(for realtime check context column), so I believe in your case it is "from-sip". This means all calls from that extension will hit this context, you can't send one call from same extension to one context and other to another, all calls will hit "from-sip" context.
I'm using asterisk with webrtc in chrome (SIPml5 client) and also using their webrtc2sip gateway.
My problem is that I can't send the hook-flash/flash signal to asterisk for some reason.
Is there a possibility to trigger hook flash by sending DTMF signals ? (which apparently work for me )
Example: I want to trigger the standard hold functionality by sending the following DTMF sequence: #123.
On the other hand as a workaround, could I hold a channel using AMI or AGI ?
Basically I want the other party to hear OnHoldMusic while the channels/call remain active. On the local side I can just mute the audio tag which is fine.
Thanks.
Yes, you can do hold using ami command hold or AGI exec command musiconhold
That info is availible in AMI/AGI documentation
You can handle dtmf by using features.conf
I would want to send a message from the server actively, such as using UDP/TCPIP to a client using an arduino. It is known that this is possible if the user has port forward the specific port to the device on local network. However I wouldn't want to have the user to port forward manually, perhaps using another protocol, will this be possible?
1 Arduino Side
I think the closest you can get to this is opening a connection to the server from the arduino, then use available to wait for the server to stream some data to the arduino. Your code will be polling the open connection, but you are avoiding all the back and forth communications to open and close the connection, passing headers back and forth etc.
2 Server Side
This means the bulk of the work will be on the server side, where you will need to manage open connections so you can instantly write to them when a user triggers some event which requires a message to be pushed to the arduino. How to do this varies a bit depending on what type of server application you are running.
2.1 Node.js "walk-through" of main issues
In Node.js for example, you can res.write() on a connection, without closing it - this should give a similar effect as having an open serial connection to the arduino. That leaves you with the issue of managing the connection - should the server periodically check a database for messages for the arduino? That simply removes one link from the arduino -> server -> database polling link, so we should be able to do better.
We can attach a function triggered by the event of a message being added to the database. Node-orm2 is a database Object Relational Model driver for node.js, and it offers hooks such as afterSave and afterCreate which you can utilize for this type of thing. Depending on your application, you may be better off not using a database at all and simply using javascript objects.
The only remaining issue then, is: once the hook is activated, how do we get the correct connection into scope so we can write to it? Well you can save all the relevant data you have on the request to some global data structure, maybe a dictionary with an arduino ID as index, and in the triggered function you fetch all the data, i.e. the request context and you write to it!
See this blog post for a great example, including node.js code which manages open connections, closing them properly and clearing from memory on timeout etc.
3 Conclusion
I haven't tested this myself - but I plan to since I already have an existing application using arduino and node.js which is currently implemented using normal polling. Hopefully I will get around to it soon and return here with results.
Typically in long-polling (from what I've read) the connection is closed once data is sent back to the client (arduino), although I don't see why this would be necessary. I plan to try keeping the same connection open for multiple messages, only closing after a fixed time interval to re-establish the connection - and I hope to set this interval fairly high, 5-15 minutes maybe.
We use Pubnub to send notifications to a client web browser so a user can know immediately when they have received a "message" and stuff like that. It works great.
This seems to have the same constraints that you are looking at: No static IP, no port forwarding. User can theoretically just plug the thing in...
It looks like Pubnub has an Arduino library:
https://github.com/pubnub/arduino
I have an app that is connected to the balance through the serial port. The balance is quite large and pressing the PRINT button is not an option. So my app asks the balance to print programmatically upon a certain user action. The balance interface allows it, and defines a print command. All works for awhile. Then after weighting few items, the balance starts outputing the previous weight....I am buffled at this point since there are few commands defined and there is not too many options to what can be done. I am already flushing out the OUT buffer after each time. So I don't know why it keeps giving me the old value.
Here is my code:
if (askedToPrint)
{
_sp.DiscardOutBuffer();
//ask the balance to print
_sp.Write("P\r\n");
}
_sp - is a SerialPort object
I am using WinCE 6.0 and Compact Framework 2.0/C#
if you are reading data from serial port using Readline() or Read() then there is a possibility that balance have sent multiple packets queued. So before reading you have to discard already pending packets. other way around is before writing request to print use ReadExisting() method to read all available data. So after sending command if your balance still sending old packets then there might be a problem with balance.