I have faced this issue many times, when I call on my sip line the server responds that the line is busy, though no call is going on when I see with asterisk -vvvr command. When I reload it, it starts working.
Is there a way to troubleshoot this problem or at least get a trigger form some command that sip needs to be reloaded.
Thanks in Advance
Asterisk work for years without reload.
Try update it to latest system or find bug in your config(like no dns, nat settings changed, bad router etc).
Asterisk have no any triggers for such case. You can use external monitoring system(like nagios) which will check sip channel and reload if needed. But better find issue and fix it.
Watch the CLI with a high debug/verbosity to see what's going wrong.
Set verbosity to a high value:
core set verbose 10
Set Debug to a high value:
core set debug 10
You can also check the status of the SIP endpoint with:
sip show peer <extension> (Replace with the number you'd use to dial the endpoint).
If you add verbose CLI output and the output of sip show peer to your question we'll be able to tell you how to fix the issue. Use pastebin or a similar site if the output is very long.
Related
I have a VPN configured by ovpn in my mikrotik that already works, but whenever I connect both by android and Windows, I get the error:
ovpn,debug,error,,,,,,,,,l2tp,info,,debug,,,critical,,,,,,,,,,,,,warning duplicate packet, droppin
On console and terminal:
I have already looked for the official forum of mikrotik but neither has no clear answer or a definitive solution.
My goal is to remove this error from my terminal and from my console log.
My current setting:
OVPN SERVER:
PPP PROFILE:
#Edit:
After contacting support, I got this error message to have no impact on the VPN. Below is the email from Mikrotik Support:
Email:
"Hello,
This error message does not have any impact on the VPN connection
establishment, it simply warns you that the client sent duplicate
message which some client software (for example Windows) do.
Best regards, Emils Z."
Thank you all for your help.
I would advise that this is probably a bug in the OVPN implementation on Mikrotik's side. Please log a support ticket with them, and provide the supout.inf file as per the usual process.
Please see this link on how to make the support info file:
https://wiki.mikrotik.com/wiki/Manual:Support_Output_File
In short, open winbox, click the make support file on the left, and then go to files and download the file to your pc. Then attach this file to the support ticket.
Problem: My main issue is that when I “REGISTER” a client via Kamailio, and I attempt to “Dial” a different endpoint within an Asterisk Dial Plan, Asterisk throws an error stating that the endpoint (the number I am dialing via “Dial”) is not registered or reachable. However, commands like “Playback” do work correctly for the client I “REGISTERED” via Kamailio.
E.g. I register client 10001 in Kamailio, I then register another client 10002 in Kamalio; both 10001 and 10002 can exercise an Asterisk Dial Plan which will play videos/audio (No Problem). But, now I want 10001 to Call (Dial) 10002; it is at this point that Asterisk throws the error “10002 is not registered or reachable”.
I have tried many of the suggestion on many different help boards (several times) but I am still unable to forward a registration from Kamailio to Asterisk.
With my current Kamailio configuration (I do use dispatching), I see , via tcpdump, Asterisk receiving a “REGISTER” request, and Asterisk sends back the “unauthorized” as expected, however, Kamailio does not re-send the “REGISTER” as is customary. I am not sure of the next step to take, but I feel I have a couple of options.
- I can continue to try and figure out why Kamailio is not sending the second “REGISTER” (I have not yet been able to figure this out).
- Tell Asterisk to not require authentication. (I am using pjsip and do not know how to not require authentication in Asterisk when the request is from Kamailio).
I have put a lot of time into this one, and I am at a sticking point. Any help or suggestions would be very much appreciated.
Thank you,
Kamailio is proxy. It SHOULD NOT do send second register unless you EXPLICTLY ask it do that in dialplan.
Dispatcher module is fast processing module. It should not do for you all staff, it just give you suggestion for dispatch.
You should not do check on asterisk for registration, you have send request to kamailio, and kamailio SHOULD do that work(it do much faster and HAVE info).
Main issue is:
asterisk main goal - give easy to understand platform for begginers. So anyone can get working pbx for free. It work on top level, with calls.
kamailio main goal is PERFOMANCE. It is not for begginer, you need have solid understanding of sip protocol, not just know that you want call. You have define what to do on packets level.
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'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.