I have this FreePBX server hosted at OPL.uk - and i want to add my two sip trunk with one number on each with two lines on.
Problem 1:
I have add one SIP trunk, as a test, as a Chan_pjsip. But when i call the number/Trunk, I get a promte “Please enter you password followed by the #-key” - before that I got a “Bla bla is not availble” or something like that
I have looked at things over and over + followed a some tutorials but no luck so far.
So what have i done wrong…? or must it be added as a chan_sip?
Problem 2:
When people call one of my numbers the call to go to a specific queue with a failure to my mobile number when no ext are logged in to queue - i cant test is at time as i have problem with adding the trunks.
Problem 3:
As my call center have to queues, one for some service people and one for the clients, depening on wich type i want to call service people or clients i want to use the diffrents trunks, with the funny thing that on the trunk that i want to call clients i want to use another number.
so is the a person that can help me over skype to due a guick overview on things
My sip provider does not have any support for this PBX, i am using UNI-TEL which is a danish provider
Here’s a quick overview of the pieces that might help:
Trunks connect your PBX to a provider’s IP address. Each IP address should have one, and only one, trunk.
When calls arrive over a trunk, the Direct Inbound Number associated with the call (the number the customer dialed) is sent to the PBX. The Inbound Routes are set up based on this DID information. You can have as many DIDs as your provider is willing to send over a specific trunk, I for example, about about 25 DIDs on one of my trunks. These are handled by the Inbound Routes.
Queues are the destination of your Inbound Route. When a call comes in a specific DID, the Inbound Route will send it wherever the route is set up to send it. This is typically a queue, a ring group, or an extension (or any of the 50 other destinations available). Note that you can also send calls to a specific Inbound Route using the CID (Caller ID of the caller) which allows you send your technicians to a different queue than your customers as long as you know their Cell Phone numbers.
Queue failover is handled at the queue and is a function of setting up the destination correctly. If the queue fails and does not send your call to where you expected it to go, it is almost certainly because you’ve set up the destination wrong.
Outgoing calls and incoming calls are almost completely unrelated. Remember, in 1) above, I said that a trunk is your connection to a SIP provider. If you try to send a call to a server that doesn’t accept calls, it will not work. If you don’t authenticate with the SIP provider, your call will fail. If you don’t include the right headers, your call will fail. The logs from the system will tell you a lot about your problem. They are located at /var/log/asterisk/full.
How can your SIP provider NOT support Asterisk? It’s a rhetorical question. It’s SIP - of course they support Asterisk. What they don’t do is provide step by step setup instructions for people that don’t know what they’re doing.
Different Trunks and different outbound numbers are almost certainly unrelated. You want to specify the outbound Caller ID, which is different than what you are asking for. If your trunks support “Foreign Caller ID”, you can set your outbound CID to literally anything you want. If they don’t, you will need to set the outbound caller ID to one of your numbers.
Once again, trunks are how you connect your PBX to your provider. It has nothing to do with numbers, Caller ID, or anything else. It’s all about passwords, IP addresses, and registration strings (if your provider uses them). You need to adjust your mindset - your understanding of how the system works is different than how it actually works.