I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication. Anyone seen this? Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060410/8a43cba2/attachment.htm
Douglas Garstang wrote:> exten => 2001,1,Page(SIP/3254105) > > does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication.It's perfectly normal and expected, if you understand at all how Page() works. Page() does not set up a call between the caller and the callees; it puts them into a conference, so it has no idea if the callee's phone is still ringing or not. In fact, it wouldn't make sense for it to pay attention to that at all, given that there could be many callees; should the caller not be put into the conference until _all_ the callees have either answered or been determined to be busy/not available/etc.?
Please look at: http://www.sineapps.com/news.php?rssid=1130 SNIP...
Point taken.> -----Original Message----- > From: Kevin P. Fleming [mailto:kpfleming@digium.com] > Sent: Monday, April 10, 2006 10:17 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] App Page() in 1.2.5 > > > Douglas Garstang wrote: > > > exten => 2001,1,Page(SIP/3254105) > > > > does strange stuff. The caller's phone immediately drops > into the call, while the callee's phone is still ringing. I'd > think it was a SIP messaging issue, except that the Dial() > command is working fine, which makes me wonder if it's a bug > in the Page appplication. > > It's perfectly normal and expected, if you understand at all > how Page() > works. Page() does not set up a call between the caller and > the callees; > it puts them into a conference, so it has no idea if the > callee's phone > is still ringing or not. In fact, it wouldn't make sense for it to pay > attention to that at all, given that there could be many > callees; should > the caller not be put into the conference until _all_ the callees have > either answered or been determined to be busy/not available/etc.? > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
As to original poster of the how-to page information, I apologize for my sloppy proof-reading, or rather lack of. The word answer is mis-spelled the mis-spelling was not intentional and as a result NOT required!!!> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > kevin ling > Sent: Monday, April 10, 2006 10:40 PM > To: jnovack@stromberg-carlson.org; 'Asterisk Users Mailing > List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] App Page() in 1.2.5 > > Hi, > > // sipura spa-941 auto-answer & paging test exten => > *63,1,SIPAddHeader(Call-Info:\;answer-after=0) > exten => *63,2,Dial(SIP/203) > exten => *63,3,congestion > > I have make some test to paging extension 203. It's work on > SPA-941. Can you test on your SPA-841? > > Kevin > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > John Novack > (port) > Sent: Tuesday, April 11, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] App Page() in 1.2.5 > > kevin ling wrote: > > >Hi, > > > >It's work on my spa-941. I just add belowing line before dial the > extension. > > > >exten => s,3,SIPAddHeader(Call Info: Anwser-After=0) ; This > is for the > >Snoms and Others > > > >Kevin > > > > > > > Is the mis spelling of Answer required to make it work? > > Problem in my case is it DOES auto answer, but the phone > receiving the page also has dialtone out of the speaker, > whereas the party sending the page does not hear it. > How to make the phone NOT play dialtone? > > John Novack > > >-----Original Message----- > >From: asterisk-users-bounces@lists.digium.com > >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John > >Novack > >Sent: Tuesday, April 11, 2006 3:54 AM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: Re: [Asterisk-Users] App Page() in 1.2.5 > > > >Anyone have this working with the Sipura 841? > > > >I can page the phone, but it auto answers the page with dial tone, > >which isn't heard by the paging phone, > > > >John Novack > > > >Alexander Lopez wrote: > > > > > > > >>Please look at: > >>http://www.sineapps.com/news.php?rssid=1130 > >> > >>SNIP... > >> > >> > >> > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
It works as I described in an earlier post. The page comes through, BUT the phone also plays dial tone from the phone when it auto answers Except for the dialtone, it works Asterisk 1.2b1 and the last Sipura software version Using the Page application in 1.2.?? provides the same result with the same syntax. John Novack kevin ling wrote:>Hi, > >// sipura spa-941 auto-answer & paging test >exten => *63,1,SIPAddHeader(Call-Info:\;answer-after=0) >exten => *63,2,Dial(SIP/203) >exten => *63,3,congestion > >I have make some test to paging extension 203. It's work on SPA-941. Can you >test on your SPA-841? > >Kevin > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Novack >(port) >Sent: Tuesday, April 11, 2006 10:27 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] App Page() in 1.2.5 > >kevin ling wrote: > > > >>Hi, >> >>It's work on my spa-941. I just add belowing line before dial the >> >> >extension. > > >>exten => s,3,SIPAddHeader(Call Info: Anwser-After=0) ; This is for the >>Snoms and Others >> >>Kevin >> >> >> >> >> >Is the mis spelling of Answer required to make it work? > >Problem in my case is it DOES auto answer, but the phone receiving the page >also has dialtone out of the speaker, whereas the party sending the page >does not hear it. >How to make the phone NOT play dialtone? > >John Novack > > > >>-----Original Message----- >>From: asterisk-users-bounces@lists.digium.com >>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John >>Novack >>Sent: Tuesday, April 11, 2006 3:54 AM >>To: Asterisk Users Mailing List - Non-Commercial Discussion >>Subject: Re: [Asterisk-Users] App Page() in 1.2.5 >> >>Anyone have this working with the Sipura 841? >> >>I can page the phone, but it auto answers the page with dial tone, >>which isn't heard by the paging phone, >> >>John Novack >> >>Alexander Lopez wrote: >> >> >> >> >> >>>Please look at: >>>http://www.sineapps.com/news.php?rssid=1130 >>> >>>SNIP... >>> >>> >>> >>> >>> > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >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/20060411/b942f008/attachment.htm