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: [email protected] [mailto:asterisk-users- > [email protected]] 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: [email protected] [mailto:asterisk-users- > > [email protected]] On Behalf Of Dave Fullerton > > Sent: Monday, February 13, 2012 9:39 AM > > To: [email protected] > > 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 -- _____________________________________________________________________ -- 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
