[App_rpt-users] AllStar Link SIP Trunk in FreePBX

[mike at midnighteng.com]

James,

First it should be said that sometimes the result may be from more than
one problem.

>From what you have shared, it is still very hard to pinpoint a single
problem but I might mention a few potential ones...

If you are not totally in tune with asterisk dial plan protocol, I would
suggest not using 70 for your interactive menu (or anything else 7x) while
you have other extensions that are 7xxxx & 7xxxxx unless you are sure you
can define the sort order for asterisk to find the extension (via context
or other scheme) or it is not possible for the longer extensions to be
70xx or 70xxx. Asterisk will always take the first match it finds. Perhaps
you have considered and compensated for this. Thought I would mention it.
Without seeing your whole dial plan, I can't say it is a problem, just a
potential one to look for.

The audio problem sounds like it may be that the pi is running out of
headroom or cpu cycles to process it during the event. Possible some other
process may be soaking up either or both resources during the event to
create the error.

Just a few thoughts for you to consider.

...mike/kb8jnm

> 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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