• Mitel 3300 Sip Trunk Setup

@MatthewWay The easiest way to configure this is to use queues. You can setup queues in queues.conf. Then cascade from one queue to another.

Mitel 3300 Sip Trunk Setup

For instance, you may setup an initial queue that has a timeout of 10 seconds and only one member: the receptionist. Once that times out, the dial plan will call the next queue, which has the receptionist, and two other people in it allowing a greater chance that someone will be able to answer the call. Last, it may have another 10 seconds that it executes a third queue that rings everyone's phone, and finally terminates to voicemail on second 31. I don't see the need for follow-me or queues. It seems like a simple requirement that can be met using standard modules of FreePBX: Inbound Route: Set 'DID Number' to that of the operator; Set 'Destination' to Ring Group 1 Ring Group 1: Set 'Ring Time' to 10 seconds; Set 'Extension List' to operator's extension; Set 'Destination if no answer' to Ring Group 2; You may also want to tick the 'Skip Busy Agent' option. Ring Group 2: Set 'Ring Time' to whatever; Set 'Extension List' to all the extensions in your hunt group; Set 'Ring Strategy' to 'hunt' or 'ringall'; Set 'Destination if no answer' to whatever you want if the hunt group fails to answer.

More Sip Trunk Setup Trix Box Asterisk videos.

Magne, It sounds like you put your “register =” first in the file? You can’t do that, it has to be within the “general” section, within a context preferebly, like in my example above. Leon, The configuration for outgoing calls is the section under the “; Register and get calls from Foo Provider” comment in the sip.conf example above. The definition of “fooprovider” (an example provider) is directly underneath, in the “fooprovider” section. These are then used in extensions.conf to make outgoing calls. In my example above, any number dialed starting with a “9” and having more than 5 digits will be routed out to the provider (see the bottom of the example extensions.conf above).

Hello, I have been having some serious problem trying to get my asterisk system to register with my sip provider. Could you please help me figure out why I am not able to connect to my sip provider? general register = username:password@sip.fooprovider.com flowroute;keep this lowercase, do not change format type=friend secret=passworkd username=username host=sip.fooprovider.com dtmfmode=rfc2833 context=inbound;change to ‘ext-did’ or ‘from-trunk’ for asterisk@home canreinvite=no allow=ulaw allow=g729 insecure=port,invite fromdomain=sip.fooprovider.com Your input on this will be much appreciated. Thanks Tamaso.

This is what I get after I click on Get an Ekiga PC-to-Phone account in Ekiga. Nonetheless, I have some criedt by another SIP provider, so I tested this from Ekiga. I was able to make a call and it was not too bad but the quality was not very good (some noise introduced when somebody speaks, a bit choppy as well), although I tested several codecs. Then I tested Twinkle for the same thing and it worked great completely clean voice from both sides! I am afraid that it might be caused by ALSA if I use ALSA in Twinkle instead of OSS I have similar problems as in Ekiga which uses ALSA only. I also tried to make a video call but without success so far both sides can see the webcam works fine in Ekiga when previewing before call but inside the call, there is no picture from remote side, on both sides. Well, it is in the beta version and after these problems are solved it might be a nice piece of software indeed!

Does anybody know info on how I can have a SIP trunk (6 channels), and any incoming call on it automatically connects to another SIP trunk that also has 6 channels? I am trying to conenct an intercom system and Vocera. Both of these systems connect on SIP trunks, but I need a SIP trunk to connect to a SIP trunk. I have both trunks connected to TrixBox just fine, and I can test outbound calls to each using a softphone.

But I cannot see how to setup incoming call routing to get to another trunk?? It only allows me to send incoming calls to an extension??