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cbrogers

join:2003-12-01
Plano, TX

Asterisk@home Questions

I have built an Asterisk@Home box, and have a few questions for those experts out there. I have setup two trunks for pstn termination, two outgoing routes, and I am getting confused on the dial plans for each trunk and ougoing route. I want to be able to choose which trunk it will take. Can some please help me?
permalink · 2005-10-17 11:57:47 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:

Re: Asterisk@home Questions

How are you wanting to choose which outgoing trunk it will use? By extension or by dialed numbers??
--
FWD#: 223611
permalink · 2005-10-17 12:19:45 · Reply to this
cbrogers

join:2003-12-01
Plano, TX

Re: Asterisk@home Questions

by dialed numbers.
permalink · 2005-10-17 12:41:36 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

1 edit
Click for full size
When you setup the outbound routing, you put a number pattern in the Dial Patterns section. For toll free numbers I have:

1800NXXXXXX
1866NXXXXXX
1877NXXXXXX
1888NXXXXXX

in that section. You then define at the bottom of that screen, the trunk order in which you wish to use.

See pic above...
--
FWD#: 223611
permalink · 2005-10-17 12:47:31 · Print · Reply to this
cbrogers

join:2003-12-01
Plano, TX

Re: Asterisk@home Questions

I have that working. I have 1XXNXXXXXX for all calls outgoing. However I have two providers voipjet and voicepule connect which both require 11 digit dialing. So a@h never just picks the first trunk to dial out. So on both outbound routing I have 1XXNXXXXXX for each provider. It just uses the first one which is voipjet. Is there a way so that if i hit 9 then the number it will use voipjet and if I use 8 then the number it will use voicepulse?
permalink · 2005-10-17 12:57:30 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

 Yes, you can...

If you hover your mouse button over the "Dial Patterns" phrase, you will get an example of how you can set it up.

For what you want you would create two outbound routes.

The first for using a 9 to get to voipjet you would put in the patterns section: 9|1XXNXXXXXX

And for Voicepulse outbound you would put in: 8|1XXNXXXXXX as the pattern. the "|" character tells the system to look for the specified digit pattern, but do not include any digits to the left of of the "|" character in the number sent to the provider.

So if you dialed 91502555121 then the system would use the 9 to select that out bound route and only transmit 15055551212 to your provider.
--
FWD#: 223611
permalink · 2005-10-17 13:08:17 · Print · Reply to this
cbrogers

join:2003-12-01
Plano, TX

Re: Asterisk@home Questions

So I do not need any dial patterns on the trunks right? This is all done via the outbound routes?
permalink · 2005-10-17 13:20:57 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

Re: Asterisk@home Questions

That is the way I do all of mine and it works great. And if you have an issue where you want a specific number to go through a specic route; Say you want to dial 915025551212 to specifically go out voicepulse, but the 9 is setup for voipjet, you can create an additional outbound route with the dial pattern of 9|15025551212 and just move it above the voipjet outbound route and it will send it through voicepulse.

So I guess what I am trying to say is that the order preference is based on the order from top to bottom as listed in the routes on the right side of the page.
--
FWD#: 223611
permalink · 2005-10-17 13:26:28 · Print · Reply to this
cbrogers

join:2003-12-01
Plano, TX

Re: Asterisk@home Questions

I am still having problems with my dial plans. I have two outbound providers and do not have any dial patters in either of the trunks. I only have dial patters in the outbound routing. I want to be able to choose which outbound route based on a number. I cannot get this to work.
permalink · 2005-10-17 22:31:19 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

1 edit
Click for full size
Click for full size
Ok, lets take a step back here...

You have the trunks setup... And lets use Voicepulse as the test.

In your Voicepulse trunk setup, you have cleared out all dialing rules. At the top of the web page when you are at the setup screen for the Voicepulse trunk, it should say "In use by 1 route."

Next, look at your setup for the outbound routing you are trying to use for Voicepulse. In the dialing patterns you should have 9|1NXXNXXXXXX as the pattern. Under the trunk sequence, you should only have your Voicepulse trunk listed.

Once you have this setup this way, try dialing on your phone, 918042681212 and see if you get the weather forecast from Verizon in Richmond, VA

FYI - the | character is obtained by holding your shift key and pressing the key above the enter key that has the back slash on it.

If this does not work for you we can dig a little deeper.
--
FWD#: 223611
permalink · 2005-10-18 08:48:41 · Print · Reply to this
cbrogers

join:2003-12-01
Plano, TX

Re: Asterisk@home Questions

I finally got it working. The thing that was messing me up was the default outbound route that is there when you install asterisk@home. It had a 9| for the dial pattern. Once I deleted that, it is working. I can now dial 8 + the number and it goes out voipjet or 9 + the number and it goes out voicepulse. I got my Grandstream 2000 delivered yesterday and got it setup. Very cool phone. Thank you so much for your help.
permalink · 2005-10-18 09:41:06 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:		 Anytime... And if you need help any other time, setup freeworld dialup account and call me on 223611
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FWD#: 223611
permalink · 2005-10-18 10:05:40 · Reply to this
TemporalFlux
Premium
join:2003-08-07
Ont, Canada
·Rogers Hi-Speed
·TekSavvy Solutions..

 Wow I didn't know there was a web interface for this... I have been editing the files with VI. Can someone explain what a trunk is? I started playing with this just a few days ago...
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If you need a compiler to compile a compiler then where did the first compiler come from?
permalink · 2005-10-18 13:28:38 · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

 A trunk is an incoming/outgoing connection to the outside world. So if I have an account with Broadvoice where they send me telephone calls or I send out going calls to them, then I set them up as a trunk.

Do not get that confused with phones inside your location, as they are called extensions.
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FWD#: 223611
permalink · 2005-10-18 13:47:10 · Reply to this
TemporalFlux
Premium
join:2003-08-07
Ont, Canada
·Rogers Hi-Speed
·TekSavvy Solutions..

Re: Asterisk@home Questions

Ahhhh!! Ok I have a trunk with iax.cc and I didn't even know it I am slowly learning the terminology. Thanks for the info!

I am wondering if there are any other companies like iax.cc that may be better for me... Any suggestions? I am in Ontario.

Also I am under the impression that if I have two external SIP extensions that they can have the PBX set up the call but the audio will be sent directly between the extensions. Is this true or am I in the potato field?

Thanks, Matt
--
If you need a compiler to compile a compiler then where did the first compiler come from?
permalink · 2005-10-18 14:40:35 · Print · Reply to this

ropeguru
Premium
join:2001-01-25
Bridgeport, WV
clubs:
·VOIPo

Re: Asterisk@home Questions

said by TemporalFlux See Profile :

Ahhhh!! Ok I have a trunk with iax.cc and I didn't even know it I am slowly learning the terminology. Thanks for the info!

I am wondering if there are any other companies like iax.cc that may be better for me... Any suggestions? I am in Ontario.

Also I am under the impression that if I have two external SIP extensions that they can have the PBX set up the call but the audio will be sent directly between the extensions. Is this true or am I in the potato field?

Thanks, Matt
There are lots of carriers out there. Best thing I can say is to go over to www.voip-info.org and look around. Some great info over there.

As far as your SIP audio question. Yes, when you dial from one extension to the other, you can have your Asterisk box setup the call then let the audio go directly between the two extensions. That is call a reinvite.

If your extensions are on the same network with no NAT between it works pretty well. But SIP and NAT do not get along very well. So I just always allow the Asterisk box just pass the audio for me. If you are using the same codecs, ie. GSM, G711u or G729, on every device, then Asterisk does what is called a native bridge and doesn't utilize hardly any cpu time. Only when it has to transcode between two different codecs does it start eating at your cpu usage.
--
FWD#: 223611
permalink · 2005-10-18 14:50:04 · Print · Reply to this
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