I have tried to make calls from my same Outbound Route number to my Inbount Route.
I have Inbound Route >> IVR >> extensions.
When I call from my outbound route to Inbound Route (same number, example: 955555441 IN and OUT 955555441) it does not work, I answer the call but the IVR does not come out. The call recording is made but nothing happens.
thank you so much!!
Some providers not support SIP loop. There is some logic which eleminate loops, which do reinvite. Some switches can't send rtp to own ip.
Solution is create custom trunk type custom to Local/$NUMBER$#from-pstn/n and create outbound route just with your DID to that trunk. It will go your inbound context without hitting provider.
Related
I need to connect the called party to another destination on an Asterisk.
Simple situation:
Inbound call is coming in, answered, welcome prompt, gets connected by the DIAL command to destination 1 (agent 1st level support).
Agent 1st level has to consult agent 2nd level, while inbound call is on hold/parked. In some situations the inbound call then has to be connected to the 2nd level agent.
Any idea how I can control the call to the called party (agent 1st level)? Isn't it just a call transfer situation with a conversation between the forwarding and the second destination?
I am using phpagi so I can send all commands from php scripts - but it doesn't make any difference to dialplan commands.
Thank you for your ideas and help
Kim
You can't do that from AGI, becuase it have no control while in Dial command
You can use transfer from softphone or AMI action Redirect.
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 have a java stasis application on Asterisk 14 using ari4java. It mostly works great. I am now trying to receive an external call and relay it back out. I do following
Incoming call enters Stasis
Create bridge
Add first call(channel) to bridge
Create channel
Add second channel to bridge
Dial( secondChID, "Local/2601", 30)
No matter what I try, the second outbound call gets the callerID of the first inbound call. That is actually OK for many calls, but in this case I want to set another callerId.
Before Dial() I have tried to setChannelVar(CALLERID(num)) and this value I can see in all events coming from Asterisk. But once the SIP call is placed, no sign of my callerID.
I doubt it is the ari4java doing anything wrong as I see the callerID in all the "dial" events. I thought I could force a callerID in sip.conf, but unable to do that too.
Does Asterisk / FreePBX support the ability to pass the caller ID of an inbound caller to a remote support agent (on a cell phone)?
Our work has a queue for incoming calls which contains "remote agents" (people on cell phones). To the cell phone agents, all calls appear to be coming from our main number (385-111-1111). We would like the calls to appear to be coming from the caller (201-555-5555).
This is not a problem with our SIP trunk provider. In the past we used different PBX software, with the same SIP trunk provider, and it was able to set the Caller ID properly. Extensions are capable of setting and passing arbitrary Caller ID, only calls from queues retain the main number.
Outgoing PEER Details:
host=sip.provider.com
type=friend
trustrpid=yes
sendrpid=yes
I've manipulated so many settings that I've come to wonder if Asterisk / FreePBX simply does not support this. Has anyone successfully been able to do this?
Asterisk certainly does. Capture the CID in a dialplan variable at the beginning of the call and set the outbound CID to the same value before passing it on.
There's no direct way to do this within the FreePBX GUI but there is a workaround:
Set up a virtual extension
Enable follow-me on the extension, add the mobile number to the follow-me list
Set the follow-me CID mode to default
Ensure the queue's agent restrictions allow the use of follow-me numbers
Have the agent log into the queue using the virtual extension instead of their mobile number
The default behaviour for the follow-me extension is to pass the incoming caller ID out. So, some flexibility is lost (mobile numbers have to be changed in follow-me settings) but it does allow the desired behaviour.
Asterisk supports setting the callerid for all outgoing or redirected calls. I did this with v1.8 and v13.7 as I'm facing the exact same requirements.
This feature depends on the provider and the contract they setup with you. My Provider calls it "Special Arrangement / Clip no screening". In my case they use "P-Asserted-Identity" to find callerid.
I had to set the following options in the outgoing sip trunk in sip.conf:
trustrpid=yes
sendrpid=pai
I have a toll free DID that users call to access my PBX service on an Asterisk box. The problem is; this DID comes only with a single channel so the system can only receive one call at a time. My initial idea was to simply get the caller ID of the incoming call, disconnect the caller and issue an automated call back to him to proceed with the call. This would free up my toll free number but could be confusing for the caller of course and also, there are issues where the caller calls from behind an extension. The best solution would be to somehow seemlessly switch the call to an outgoing trunk to reconnect the caller but now using my SIP trunk.
My question is; is there a way to do this in Asterisk (or I guess, does SIP somehow allow such operation)?
Thanks in advance.
That is called "callback".
Yes, you can do it. No, asterisk have no internal way do that and no way do it not noticable for user.