I have been tried couple of hours to outbound call from Asterisk using dongle. But it shows following error
WARNING[22180]: channel.c:166 channel_request: [dongle0] Request to call on device which can not make call at this moment
[Apr ] WARNING[22180]: app_dial.c:2277 dial_exec_full: Unable to create channel of type 'dongle' (cause 44 - Requested channel not available)
Dialplan command is :
exten => _X.,n,Dial(dongle/dongle0/+73432823232)
Could you please give me solution?
UPDATE : On asterisk*CLI> i can make outboud call using this dongle cmd dongle0 ATD+434343434232;
Error mean you not configured it correctly or device turned off/unplugged
For more info enable debug or do
dongle show devices
https://github.com/bg111/asterisk-chan-dongle
Related
My goal
Pass an incoming call directly to Stasis and allow the app to decide whether to play a ringing or busy tone to the caller.
The problem
With my ARI app, if I omit the same => n,Ringing line from my dialplan, the Stasis app returns an error if the caller hangs up. I can have a call hang without any early media, until I pass a channel.play() command, via the ARI.
This solution has 2 issues:
The Stasis app receives a second StasisStart when the caller hangs up, returning a Channel not found error.
There is no command for channel.busy
Does anyone have any suggestions?
My only option that I can currently see is to ensure that all users have voicemail and a busy tone is never played. Not everyone wants / likes voicemail and it is not ethical to answer the call and play a busy tone, without the caller knowing that their call is connected.
Update
Using the following dialplan, I can get this to work in the desired way (plays busy to a user if they're not available), but with an error:
extensions.conf
[public]
exten => _.,1,NoOp()
same => n,Stasis(myStasisApp, ${SIP_HEADER(to)})
same => n,Busy(10)
same => n,Hangup()
myApp.js
// The user is available
channel.ring();
// The user is busy
channel.continueInDialplan();
Error
Another StasisStart is sent when the caller hangs up, followed by:
Unhandled rejection Error: {
"message": "Channel not found"
}
We faced the same problem and lost precious time finding out the reason so I'm sharing the solution here and maybe it will help.
extensions.conf
[public]
exten => _.,1,NoOp()
same => n,Stasis(myStasisApp)
same => n,Hangup()
When Asterisk receive a call, it will start the stasis app.
Create a Bridge.
Add the incomming channel A in this bridge.
Create a new outgoing channel B from your ari app with POST /channels/create.
Add the outgoing channel B in that bridge.
Dial from channel B the destination, where both of the channels
are in the same bridge with POST /channels/{channelId}/dial
Now, you will be able to see a new ARI Dial events with Ringing and Answer.
For the Hangup, you will receive Channels end events with Hangup Cause Code not a text like 17 for busy
Asterisk Hangup Cause Mappings
It's simpler to originate a channel (Asterisk version 13) instead of create and dial (Asterisk version 14) but you will not have the early media or a full control on that channel because it's created by Asterisk and not the ARI app so this channel will start sending event back to ARI when the call start and not before.
Asterisk 14 ARI: Create, Bridge, Dial.
ARI and Channels: Manipulating Channel State
This thread helped us a lot:
Re: ARI: add channel to bridge immediatelly after originate action
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Have fun ! and Hope this will help.
/ohammami
I got the above error when try to connect two soft phones which is successfully registered. I was trying to make a voice call in local not connecting and instead returns the error:
chan_sip.c:4274 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
I am using asterisk 13.5.0 and not using freepbx. Simply try to make a call between two peers
Make changes to sip.conf edit the line bind address like bindaddr=0.0.0.0
Likly firewall or permissions(for socket) issues.
Solved
OK I've just solved this issue in my network.
First of all , about the situation : i had an "Issabel VoIP server" based on Asterisk 13 but when i wanted to make calls ,it just failed and i got the same error as yours on Asterisk CLI. I had successful ping requests from the server and extension had been registered without any errors.
Solution: I had doubt about network devices, so i installed 2 soft-phones (like 3cx or Zoiper) on my laptop and connected it straightly to the server using a LAN patch cord ( to see if there's an issue on server side or its related to my network infra.) and registered 2 extensions on my soft-phone and they called each other without problem.so i understood that its one of my network devices that was limiting the transition of VoIP packets between server and extensions . finally i got that it was my TDD-LTE Modem that intercept my connection.
So to make long story short :
1- First create a small network between your server and laptop
2- Register 2 extensions on your soft-phone
3- make call between them and if it goes fine , then look for an obstacle in your main network.
hope it could help you
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 getting a weird behavior on my Asterisk.
Calls are dropped after 15mn.
I'm getting the following error on the log file:
NOTICE[6301] chan_sip.c: Failed to authenticate on INVITE to '<sip:41907736445#188.32.64.1>;tag=ef7143klc9'
I'm using Asterisk Realtime. Calls a received from an operator and forwarded to external numbers throught an outbound trunk provided by anther operator.
Thanks in advance
In your sip.conf, try setting qualify=yes or keepalive=yes globally or for the trunk. Make sure to reload or just restart the service.
What version of Asterisk are you running?
Some carriers may send "confirmation" invite every X minutes.
You can see more by enable sip debug
Also if exactly at 15 min calls get dropped, then i would check the firewall to see whether there is timer set there that closes the connection after 15 min. I am speaking from experience.
I'm a veteran of Asterisk 1.4 and am looking to build a new application on Asterisk 11 (which is currently beta, but is planned to be LTS release some time before I need it.)
I can't get Asterisk Manager Interface on 11 to send me any events. (Now, obviously, in production, I need to cut down these AMI rights drastically, but as I'm exploring I've opened the firehose, if you will.)
manager.conf looks like this:
[general]
enabled = yes
port = 5038
bindaddr = 127.0.0.1
[manager]
secret = squirrel
deny = 0.0.0.0/0.0.0.0
permit = 127.0.0.1/255.0.0.0
read = all
write = all
I then use telnet to try to get in and explore the event stream:
$ telnet localhost 5038
Trying ::1...
telnet: connect to address ::1: Connection refused
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.3
Action: Login
Username: manager
Secret: squirrel
Events: on
Response: Success
Message: Authentication accepted
Event: FullyBooted
Privilege: system,all
Status: Fully Booted
...and there it sits, not moving, no matter what I actually do with the system. I've also tried using the Event manager action with EventMask: on to try to get something out of it; the command is accepted, but nothing changes. It will happily respond to any other actions I send it, though.
Any leads? This sort of thing worked fine under 1.4, and I'm not finding anything in any documentation suggesting I'm doing something wrong. I suppose the next thing to try is 1.8...
(There is little else in /etc/asterisk; I'm using example configs only for reference. This is as minimal as we get...)
It's may be bug in Asteriks / FreePBX. I had same situation, and my API php script didn't receive any events from AMI.
For fix this bug, you must install "Conferences" module and restart Asterisk from SSH: service asterisk restart
I just tested this with the latest 11 from subversion using your configs. I see events being generated. For example, executing this from the CLI:
*CLI> channel originate Local/Foo application Bar
While invalid, will cause some events to be spit out to the manager interface.