[asterisk-users] Polycom firmware 4.0.1 and paging
Mike
list at net-wall.com
Mon Feb 13 09:16:49 CST 2012
Thanks Dave, it at least gives me hope that my efforts aren`t wasted.
Mike
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Dave Fullerton
> Sent: Monday, February 13, 2012 9:39 AM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
>
> On 02/10/2012 05:30 PM, Mike wrote:
> > Hi,
> >
> > I just moved many Polycom phones from firmware v3 to 4.0.1b.
> > Anto-Answer simply stopped functioning. I can downgrade and make it
> > work, upgrading kills it again. There obviously is a difference in how
> > the newer firmware is treating this auto answer sip header.
> >
> > Can anybody tell me if they have Polycom firmware 4.x.x working with
> > auto-answer/paging? Just so I know it's worth my time to investigate,
> > as opposed to knowing it`s a Polycom firmware bug? If so, did you have
> > to make any changes to the SIP header sent to make Polycom phones auto
> answer?
> >
>
> I would second the others suggestions about rewriting the configs.
> Polycom made extensive changes between 3.2 and 3.3, and I think they made
> a fair number of changes between 3.3 and 4.0. I have two phones that I've
> upgraded to 4.0.1b for testing, a 550 and a spectralink 8440, and I
> believe I have auto answer working as you describe. Here's the pertinent
> snippet from my config:
>
> <polycomConfig>
> <voIpProt>
> <voIpProt.SIP>
> <voIpProt.SIP.alertInfo
> voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
> voIpProt.SIP.alertInfo.1.value="intercom"
> voIpProt.SIP.alertInfo.2.class="ringAnswerMute"
> voIpProt.SIP.alertInfo.2.value="page"
> voIpProt.SIP.alertInfo.3.class="autoAnswer"
> voIpProt.SIP.alertInfo.3.value="silentanswer">
> </voIpProt.SIP.alertInfo>
> </voIpProt.SIP>
> </voIpProt>
> </polycomConfig>
>
> I have also added an <se.rt> section to adjust the ringer and timeouts for
> these ring tones.
>
> -Dave
>
> --
> _____________________________________________________________________
> -- 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
More information about the asterisk-users
mailing list