[kwlug-disc] Troubleshooting Asterisk disconnects

L.D. Paniak ldpaniak at fourpisolutions.com
Fri Mar 23 16:16:59 EDT 2012

Sounds like the ultimate VoIP project killer:  a bug in which a human
voice can trigger DTMF tones.  It is triggered more commonly by female
voices and non-English languages like French.  eg.


I seem to have fixed it with a firmware upgrade and by turning down the
pickup gain on the SPA.

On Fri, 2012-03-23 at 16:04 -0400, Chris Irwin wrote:
> I know there are more than a few asterisk users here.
> I'm partially switched over to VOIP. The plan was to become familiar
> with my chosen software & hardware, then migrate the landline number
> to a VOIP provider. Unfortunately, I've been having some WAF issues
> that have prevented me from switching completely. Here is my current
> setup:
> - 1x Linksys-branded SPA3102 with FXO and FXS ports.
> - 1x Askozia asterisk server
> - 1x normal cordless phone connected to the SPA3102
> - 1x service through landline via SPA3102 (this is the primary line
> due to it being "our" phone number), accessed as SIP in asterisk.
> - 1x service through unlimitel, accessed as AIX in asterisk.
> However, occasionally calls will disconnect, usually with an
> ear-piercing chirping noise. I have not observed any issues with the
> unlimitel side, leading me to believe it is a physical line or SPA3102
> issue. However, my inbound calls are primarily on the landline number,
> so there is a significant difference in call volume and usage time, so
> I can not definitively say that.
> Plugging directly into the landline appears to avoid the issue, thus
> (probably?) eliminating the line and the phone themselves as the
> issue. That leaves asterisk, the hardware it runs on, the SPA3102, and
> potential mis-configuration as issues.
> I moved my askozia server from the dedicated Wyse 3150 to a VM, as
> that was an easy quick step (which also took care of a software update
> at the same time), but that appears to have not solved anything.
> Before I start any big changes, I thought I'd reach out for help due
> to my limited experience.
> Is there an easy way to log where a call terminates from? Is there any
> information I can pull from the 3102 that might be helpful?
> Unfortunately I can't predict when a call will disconnect, so logging
> seems to be the best route.

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