[App_rpt-users] AllStar Link SIP Trunk in FreePBX

[Doug Crompton]

James,

 Here is what I have to dial sip to Allstarlink....

In sip.conf

[allstar]
type=peer
host=sip.allstarlink.org
username=1
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
In extensions....

exten => 549,1,Dial(SIP/@allstar,120,WT)
exten => 549,2,Hangup()

This is in my PBX system. When I dial 549 it rings and I get the voice prompts. You have enter several things node number, mode, pin number.   This works fine for me but I am using standard old version Asterisk for my PBX on a PC.

Are you trying to automatically enter the node, etc to the attendant? If so it might be a timing issue. I have never used FreePBX but I would think you could open a ssh session to a Linux prompt and enter the Asterisk client and be able to see some info that might be helpful during call progress.

73 Doug
WA3DSP
http://www.crompton.com/hamradio


From: james at lyljtlabs.com
To: doug at crompton.com
CC: app_rpt-users at ohnosec.org
Date: Tue, 17 Feb 2015 23:58:23 -0500
Subject: RE: [App_rpt-users] AllStar Link SIP Trunk in FreePBX

Doug, This is an Asterisk/FreePBX server on my end. And I’m trying to connect into the AllStar Link network’s direct SIP/public telephone portal (https://allstarlink.org/support.html#telephoneportal) not another node I have control over. I don’t have any access to the AllStar Link network side of this call. The public access portal should let me dial node numbers once I connect. But I’m just dialing in as a user. I might be configuring my Asterisk box wrong and shouldn’t be using a trunk? Thanks,James  		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.keekles.org/pipermail/app_rpt-users/attachments/20150218/974629aa/attachment.html>