[asterisk-users] Polycom firmware 4.0.1 and paging

Gord Urquhart gordurq at gmail.com
Wed Feb 15 17:00:00 CST 2012


It appears you need the "info=" if the string you are using is enclosed in
angle brackets.
   Alert-Info: foo    works
   Alert-Info:<foo> does not work
   Alert-Info:info=<foo> works



On Wed, Feb 15, 2012 at 2:09 PM, Mike <list at net-wall.com> wrote:

> With Polycom firmware 4.0.1b?
>
> I have 1.8, one of the latest can`t remember which is installed on that
> server. Maybe the fact that my alert info has two words, and isn`t parsed
> correctly by Polycom...?
>
>
> 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: Wednesday, February 15, 2012 10:20 AM
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> >
> > Which version of asterisk are you using? I just have this in 1.4 and it
> > works fine:
> >
> > SIPAddHeader(Alert-Info: intercom);
> >
> > -Dave
> >
> > On 02/14/2012 08:10 PM, Mike wrote:
> > > In case anybody was following this thread, or someone Googles it in
> > > the future, here is the solution:
> > >
> > > This worked fine with Polycom firmware 3.3x:
> > > exten =>  s,n,SIPAddHeader(Alert-Info:<Ring Answer>)
> > >
> > > For firmware 4.0+, apparently I needed to add info=, i.e.:
> > > exten =>  s,n,SIPAddHeader(Alert-Info: info=<Ring Answer>)
> > >
> > > Simple, yet quite obscure (for me at least).
> > >
> > >
> > > Mike
> > >
> > >> -----Original Message-----
> > >> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> > >> bounces at lists.digium.com] On Behalf Of Mike
> > >> Sent: Monday, February 13, 2012 10:17 AM
> > >> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > >> Subject: Re: [asterisk-users] Polycom firmware 4.0.1 and paging
> > >>
> > >> 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
>
>
> --
> _____________________________________________________________________
> -- 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120215/6dd80f9d/attachment.htm>


More information about the asterisk-users mailing list