[App_rpt-users] AllStar Link SIP Trunk in FreePBX

[Doug Crompton]

James,

 You didn't mention if you have the Asterisk client running at both ends to check for errors??? If  not that could be helpful.
73 Doug
WA3DSP
http://www.crompton.com/hamradio


> From: james at lyljtlabs.com
> To: app_rpt-users at ohnosec.org
> Date: Tue, 17 Feb 2015 22:59:56 -0500
> Subject: [App_rpt-users] AllStar Link SIP Trunk in FreePBX
> 
> I have a raspberry pi running FreePBX/Asterisk with a VOIP mesh network. The internal extensions and Google voice trunk are working properly. I have an outgoing dial plan configured so that 70 (should go to interactive menu), 7xxxx or 7xxxxx (xxxx and xxxxx are node numbers) will dial the AllStar Link SIP trunk. The outgoing calls will ring the AllStar trunk and often, but not always, the call connects. If it connects, it usually does not have received audio. If there is audio, then when I enter node numbers, etc it appears the DTMF tones are not recognized. I say this because every node number I enter is "invalid".
> 
> I can dial the AllStar access number, 763-230-0000, from the same phone over the Google voice trunk and everything works fine. I can also connect to the AllStar Link SIP directly using a software phone on my same LAN and it works fine.
> 
> I'm assuming I am doing something wrong in the trunk configuration. Any help would be great. Thanks- James
> ---------------------
> I have it configured in FreePBX as follows:
> Trunk Name: AllStar Link
> 
> PEER Details:
> username=1
> fromuser=1
> secret=allstar
> host=sip.allstarlink.org
> fromdomain=sip.allstarlink.org
> type=friend
> context=from-trunk
> insecure=port,invite
> trustrpid=yes
> directmedia=no
> keepalive=45
> nat=yes
> dtmfmode=rfc2833
> disallow=all
> allow=g711&ulaw
> 
> USER Context: (no entry)
> USER Details: (no entry)
> Register String: 1:allstar at sip.allstarlink.org
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