I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried the monitoring app, but that only seems to log to files. I couldn't figure out any way to set up a meetme conference while still running through the various stages in answering the call. Any ideas? Things I've considered, but maybe didn't drill down far enough: - Unload chan_oss, and get the monitoring app to just write to /dev/dsp. Or maybe to a pipe, which I can then pass through the standard linux "play" application. res_listen anyone? - Figure out meetme for this purpose. - Three way call to the console? I couldn't figure this one out either. Any of these solutions bring up the related question - is there any way to get Record to hang up automatically if any other (non-asterisk connected) extension is picked up in the house? Traditional answering machines detect this, but I'm not sure how. The easy workaround is simply to hit the # key on the local extension to force a hangup. I'd appreciate any insights (or code!). Thanks! Joel
If you're using zap channels you could use the Zapbarge app. Just set up an extension to start the app, like exten => 8000,1,ZapBarge exten => 8000,2,Hangup Asterisk then will prompt for a chan number (zaptel only!) and let you listen to what passes on that channel... You can barge from any channel, but only into zaptel (ie you can listen from sip to a zap channel). Matteo Il dom, 2003-04-20 alle 01:07, Joel Scotkin ha scritto:> I've set up asterisk with my X100P as a home answering machine. Works great > so far - answers the phone after 20 seconds, runs the phone tree, emails > voicemail, etc. > > However, the one feature traditional answering machines have that I haven't > been able to figure out is how to listen in on the call. Ideally I could > just route through Console/dsp and hear it on my speakers. I've tried the > monitoring app, but that only seems to log to files. I couldn't figure out > any way to set up a meetme conference while still running through the > various stages in answering the call. Any ideas? > > Things I've considered, but maybe didn't drill down far enough: > > - Unload chan_oss, and get the monitoring app to just write to /dev/dsp. Or > maybe to a pipe, which I can then pass through the standard linux "play" > application. res_listen anyone? > > - Figure out meetme for this purpose. > > - Three way call to the console? I couldn't figure this one out either. > > Any of these solutions bring up the related question - is there any way to > get Record to hang up automatically if any other (non-asterisk connected) > extension is picked up in the house? Traditional answering machines detect > this, but I'm not sure how. The easy workaround is simply to hit the # key > on the local extension to force a hangup. > > I'd appreciate any insights (or code!). Thanks! > > Joel > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users
Has anyone else wondered why we can't do a SIPBarge, H323Barge, etc. I took an extremely brief look at the code and it looks like it's probably zap specific - but is a barge app for the other interfaces something anyone else is writing? Steve Radich BitShop, Inc. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Sunday, April 20, 2003 5:02 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Call screening If you're using zap channels you could use the Zapbarge app. Just set up an extension to start the app, like exten => 8000,1,ZapBarge exten => 8000,2,Hangup Asterisk then will prompt for a chan number (zaptel only!) and let you listen to what passes on that channel... You can barge from any channel, but only into zaptel (ie you can listen from sip to a zap channel). Matteo Il dom, 2003-04-20 alle 01:07, Joel Scotkin ha scritto:> I've set up asterisk with my X100P as a home answering machine. Worksgreat> so far - answers the phone after 20 seconds, runs the phone tree, emails > voicemail, etc. > > However, the one feature traditional answering machines have that Ihaven't> been able to figure out is how to listen in on the call. Ideally I could > just route through Console/dsp and hear it on my speakers. I've tried the > monitoring app, but that only seems to log to files. I couldn't figureout> any way to set up a meetme conference while still running through the > various stages in answering the call. Any ideas? > > Things I've considered, but maybe didn't drill down far enough: > > - Unload chan_oss, and get the monitoring app to just write to /dev/dsp.Or> maybe to a pipe, which I can then pass through the standard linux "play" > application. res_listen anyone? > > - Figure out meetme for this purpose. > > - Three way call to the console? I couldn't figure this one out either. > > Any of these solutions bring up the related question - is there any way to > get Record to hang up automatically if any other (non-asterisk connected) > extension is picked up in the house? Traditional answering machinesdetect> this, but I'm not sure how. The easy workaround is simply to hit the #key> on the local extension to force a hangup. > > I'd appreciate any insights (or code!). Thanks! > > Joel > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all. Is there a way to use asterisk for call screening? Meaning, a call comes in, asterisk answers with voicemail after I don't pickup, and the voicemail prompt + the caller's message a played via the sound card on asterisk. If I wan't to pick up, I do so by picking up the phone and dialing something. Is it doable? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200
Sorry, I don't follow. Dialing *98 will achieve what? Up until the time I decide to take the call, I want to be able to hear the person leaving a message interactively, so when I decide to pick up the call he's still there. Like a regular answering machine> -----Original Message----- > From: hadi [mailto:mesbah@karbeh.com] > Sent: Sunday, December 19, 2004 12:13 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] call screening > > Yes > U can do it with asterisk and by dialing *98 on your Ip Phone you can > listen > to your message > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ShovalTomer> Sent: Sunday, December 19, 2004 1:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] call screening > > Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after Idon't> pickup, and the voicemail prompt + the caller's message a played viathe> sound card on asterisk. If I wan't to pick up, I do so by picking upthe> phone and dialing something. > Is it doable? > > Shoval Tomer, > IT Manager, > SofTov Advanced Systems, Ltd. > Office: +972-3-9230686 ext. 179 > Fax: +972-3-9216642 > Mobile: +972-54-8000200 > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support.
Hello-- I've done some coding for call screening in Asterisk. It's not in Asterisk yet, mainly because we're waiting for prompts from Allyson so it sounds like the rest of the system. But patches, prototype sound files, etc, are all filed at: http://bugs.digium.com/bug_view_page.php?bug_id=0000752 And I'd love to have your feedback. murf> Hi all. > > Is there a way to use asterisk for call screening? > > Meaning, a call comes in, asterisk answers with voicemail after I > don't > pickup, and the voicemail prompt + the caller's message a played via > the > sound card on asterisk. If I wan't to pick up, I do so by picking up > the > phone and dialing something. > Is it doable? > > Shoval Tomer, >
I would like to apply the app_dial macro patch referenced in: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=2905 To my stable version of Asterisk: Asterisk CVS-v1-0-12/21/04-14:14:46 built by root@S-ATL-PB01 on a i686 running Linux. Mantis has 5 attached patch files. It looks like the first two are for app_dial and the next three are for pbx.c. Can someone please tell me am I supposed to apply all of these? Just the last one? Thanks. Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
Adam Robins wrote:> I would like to apply the app_dial macro patch referenced in: > > http://bugs.digium.com/bug_view_advanced_page.php?bug_id=2905 > > To my stable version of Asterisk: > > Asterisk CVS-v1-0-12/21/04-14:14:46 built by root@S-ATL-PB01 on a i686 > running Linux. > > Mantis has 5 attached patch files. It looks like the first two are for > app_dial and the next three are for pbx.c. > > Can someone please tell me am I supposed to apply all of these? Just > the last one? Thanks.The patches are for CVS-HEAD. They will not work with STABLE. You could read the patches and the source code and try to manually merge the changes, although backporting it would probably require at least a little C knowledge. The other option would be to run CVS-HEAD. This would however not be recommended for a production system. -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)