ISN: Free SIP Dialing from * Phones

wardmundy

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Introducing ISN: Free SIP Dialing From Any Asterisk Phone

gotisn5.gif

 
Swamped with volunteers. Guess we have to work on a directory now... or maybe just an AsteriDex enhancement. :smile5:
 
I have it setup but get this in the CLI when I dial:

Executing [s@macro-dialout-trunk:26] Dial("SIP/20-b7b1a1f0", "local/1234*1061@freenum|300|") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
 
Egg-on-Face :(

I had alot of putty windows open, and, duh, made the extensions_custom.conf changes to the wrong server. Ugh.

Sorry for the noise....
 
New Conference Bridge

We're putting in place a new conference bridge for registered users on the forum. Connection details are attached. Give it a try...
 

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What would be a proper caller ID for the freenum trunk?

The * is not allowed in the CLID field.
Should we present <exten@ITAD> instead of <exten*ITAD>?
 
We need to get that fixed in FreePBX. Asterisks are permissible in a CallerID.
 
I was reading in the freenum website that they are suggesting to use a real sip uri CLID instead of the ISN*ITAD to help to a transition to full sip uri support.

I understand their goal, but it seams unrealistic as someone need to be able to call you back.
 
Thanks, Ward our number works great.
I recived our own isn number 1072 about 5 days ago but have not recived info on logging in at freenum.org, were you able to program your numbers another way. Has any one eles been able to program there own numbers.


Gary
 
I got mine earlier today. It seems that their automated system was a little bit behind.
 
I got mine today to, I entered my FQDN as per nerd vittles and await, I quess there must be a backlog from what I hope is a large demmand, anyway I will report back when it starts working.

Gary
 
Got my own ITAD today and everything works great! I opted to re-delegate my ITAD zone to my own nameserver for greater control and flexibility.

I have setup a test ISN#. You may dial to test your outbound. ISN:8378*1085 (test*1085).

test-isn.jpg
 
Hi, I followed Ward's article & opted for the first choice, setting up a trunk & OBR, and using his modified outbound route dial pattern. When I dial without the leading **, I get a dial tone. When I dial with the leading **, I get a "call failed error", and the logs are showing "No target channel found for 1234*1061.". If I just use **|X.*X. in my outbound route dialplan, I get "call failed" and see "No target channel found for 1234*1061." in the logs when dialing with the leading **

I'm in Spain - would I need to modify these instructions for European dialing?
 
Spain shouldn't matter. This is pure SIP. What type phones?
 
If you're just getting a dial tone, then the local dial plan on your Aastra phone is the problem. Try clearing it out and see if it works. Otherwise, you may need to upgrade the firmware on the phone.
 
The phone's firmware is the latest (2.5.1.2000). The only dialplan I have at this point is to dial 9 for an outside line. Since this isn't yet a production system, I'll get rid of it & let you know how it goes. Thanks for the 2 minute responses, BTW!!
 

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