[asterisk-users] Paging Application - Polycom 601

Bill Andersen andersen at mwdental.com
Thu Aug 9 09:33:29 CDT 2007


> > Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies
> >
> > We have an installation of 35 SIP phones (Polycom 501) and
> > one receptionist phone (Polycom 601).  I have 15 of the 501s
> > set up to accept a "Page".  From what I understand, the "Page"
> > is done using the asterisk page application that throws the
> > extensions into a conference room and then set the originating
> > caller to the only one who can talk.
>
> I would be curious to see how you set up the phones to accept paging,
> just to make sure there isn't something iffy with your phone
> configuration.

I'm trying to get more info on how the phones are set up.  This is
a commercial product and I have a web GUI to enable/disable paging.
What that actually does?  I don't know yet.  I'll find out.

> > The problem I am having is about 1 out of 25 pages will crash
> > the Polycom 601 (receptionist) and the phone will reboot.
>
> Is the 601 calling the page, or receiving a page from another phone?

The 601 is Calling the page.  (601 is Receptionist)

> >  This
> > leaves all the extensions in the conference room and each
> > party must hit "end call" on their phone to get out of the
> > conference.  However, the receptionist can't do that because
> > that phone restarts.  Once it has rebooted, it does not show
> > to be connected to the conference room.  However, I feel like
> > it is still "in the conference" - with no way out.
>
> You "feel" like it? Do you know for sure?

OK, now I know for sure... Had the 601 crash again this morning
and I used your help in see the meetme info.  This is roughly
20 minutes after the 601 crashed...

Conf Num       Parties        Marked     Activity  Creation
1913938683d    0006           0001       00:19:07  Dynamic
* Total number of MeetMe users: 6

User #: 01   9403225392 Reception Channel: SIP/7110-b2e11758  (unmonitored)
User #: 03         7137 <no name> Channel: SIP/7137-b2f63e80  (Listen only)
(unmonitored)
User #: 05         7129 <no name> Channel: SIP/7129-b2ca1c78  (Listen only)
(unmonitored)
User #: 09         7121 <no name> Channel: SIP/7121-b2c6a0e0  (Listen only)
(unmonitored)
User #: 15         7117 <no name> Channel: SIP/7117-0855e960  (Listen only)
(unmonitored)
User #: 20         7136 <no name> Channel: SIP/7136-b2f09b58  (Listen only)
(unmonitored)
6 users in that conference.

> If the phone does not show an active call, it's not connected to
> anything. I don't see how it would be in a conference after a reboot.
> Your problems below are probably caused by something else. The
> spontaneous reboot is telling.

I appears it is still in the conference, even after reboot.

> > After one of these crashes, the 601 phone will start having one
> > way audio (can't hear caller), various other weirdness (side
> > car status wrong) and the only way to completely correct the
> > problems are to restart asterisk - which I assume kills the
> > "rogue" page application.
>
> The 601s with sidecars have been problematic.

I'm finding that out the hard way!

> What Polycom firmware are you using?

1.6.7.0098

> > 1) Has anyone ever seen this problem?
>
> Other users have reported problems with 601s crashing. Check your
> firmware. AFAIK, the current firmware is 2.1.3.

My vendor tried to move to a 2.x firmware, but it had a real bad
delay when reading keys.  It would miss about ever 3rd or 4th key
you pressed.  Sometimes, the keys would "stick" and you'd hear the
touchtone for 10 seconds or so.  They had me move back to 1.6.7 and
it all went away...

> > 2) Is there a way from the CLI to show and kill a page?
>
> 'show channels' will show you active calls (in 1.2; in 1.4, use 'core
> show channels')
>
> 'meetme kick' lets you kick channels/users from a conference.

Thanks.  Helped alot.

> Still, I don't think that's what's happening here.

I'm no so sure.  The "one way audio" seems to show it's face
within an hour or so after a "page" that crashes the 601.
I "kicked" everyone off the meetme this time within 20 minutes
and it's been 2 hours now.  No "one way audio" yet...

Thanks for the help.

Bill




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