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@sip.allstarlink.org _______________________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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