I would suggest starting with a packet capture of the SIP messages that
will include both call legs (i.e. capture at the Asterisk box). This
should tell you who initiated the hangup - the carrier side, the phone
side, or Asterisk.
On Wed, Mar 26, 2014 at 11:46 AM, Mike Diehl <mdiehlenator at gmail.com>
wrote:
> Hi all,
>
> I have a user who is reporting dropped calls at his site. We don't
have
> any other users complaining of this.
>
> So far, this is what we know:
>
> 1. The manager bought all new Polycom phones. (POE)
>
> 2. They replaced the network switch with a POE version.
>
> 3. It's not just one or two of the phones that have problems.
>
> 4. It doesn't matter if they use the headset or the cordless set.
>
> 5. The ISP reports a very clean circuit. (Ethernet from the CLEC.)
>
> 6. We don't see their phones become unavailable very often.
>
> 7. They are the only site that seems to be having trouble.
>
> So, where else can/should I look?
>
> Mike.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
[image: Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 317 507 4029
Check us out at: http://digium.com ? http://asterisk.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140326/1dafcbc8/attachment.html>