Is there any way to stop duplicate INVITE in asterisk SIP message a?
We are using Asterisk 1.8.10
Inbound Call flow to VoIP/SIP is as follows:-
SIP trunk -> Asterisk 1.8 -> VOIP app
When we get call from SIP provide/trunk asterisk generate duplicate 2 SIP INVITE messages to VOIP app unable to handle it. Is there any way to stop it from asterisk side.
Any help will be appreciated!
Thanks
Asterisk send second invite only if
1) first invite was not confirmed. You app should answer OK(confirm).
2) reinvite requested.
You can try increase t1/t2 timer to increase delay in 1). For disable reinvite, see directmedia option.
Related
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.
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 am building an application in Asterisk , which has to simulate the phone keypress action i.e DTMF. For instance , if a caller calls a number and the IVR requires to press a digit , the application (say a webpage) should also be able to simulate the same key press. I tried the following:
- I have a web application that triggers PlayDTMF action over AMI to simulate the key press on the caller channel. Unfortunately this didn't work , as it played the DTMF digit on the channel (the keypress could be heard) but asterisk didn't receive the DTMF. It was just playing the DTMF, which is what the AMI action was designed for. I technically need to "send" the DTMF digit on the channel.
- the sendDTMF function in asterisk couldn't be used either,as it is a dialplan function and I need a AMI action to trigger DTMF from an external source.
A similar issue was reported sometime back: Asterisk AMI: DTMF not received on SIP channel
I am using Asterisk 11. Is there a way to implement the above using AMI/AGI? Or do i need to further upgrade my Asterisk to 12 to make use of ARI?
Appreciate your inputs.
I managed to simulate the DTMF action , by Dialing/Originating out to a local context and using the outgoing bridging channel id for DTMF input. So the flow is :
call lands on Asterisk -> Asterisk Dials out to a Local context -> The dialout creates a bridging event which has the outgoing channel id(store in some DB) -> The local context can have the IVR flow and using the stored channel and PLayDTMF action - dtmf digits can be inserted. Not sure if there is a more elegant solution to this!
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.
Goal is to get SIP response codes(like 100, 180, 200..) from asterisk manager interface:
What I have done:
-create new user in manager.conf in /etc/asterisk
-for getting the sip messages, connected to default port 5038 using telnet and by using 'Login' action with appropriate username, password
details successfully connects to asterisk manager interface
But in these bulk of details I am not getting actual sip response codes, i want to monitor them.
-I read different actions synopsis from asterisk cli by firing 'manager show commands' but not able to get clear idea of which action
would be useful.
How can i get those resoponse codes? Any thought would be helpful on this one.
Thanks.
Edit: Got this link while mining, http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState
this is good, but i want exact sip response code
There are no any code like that in chan_sip.c
You are welcome to add more events inside that file and rebuild asterisk.
However if you expect low-level control over sip messages, asterisk is bad choice. Use opensips or kamailio projects.