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@lyljtlabs.com
To: doug@crompton.com
CC: app_rpt-users@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