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 <mdiehlena...@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
-- _____________________________________________________________________ -- 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
