[asterisk-users] Detect DTMF tone during call?
Hi, I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked briefly through the archive and I heard mentions of AGI, is this what must be used to accomplish this? Thanks, Dan -- _ -- 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
Re: [asterisk-users] Detect DTMF tone during call?
That works perfectly! Just for anyone else who stumbles upon this the exact things I did were: features.conf testfeature = *,peer,System,/bin/touch /tmp/buzzthemin extension.ael Set(__DYNAMIC_FEATURES=testfeature); Thanks again! On Sat, Feb 26, 2011 at 3:54 AM, Roger Burton West ro...@firedrake.org wrote: On Sat, Feb 26, 2011 at 03:08:02AM -0600, Dan Saul wrote: I am attempting to create a intercom buzzer system using asterisk as a back end. Most is figured out except the actual action of buzzing the door. I need to detect whether a DTMF key was pressed by the the called party (the resident). Is this possible to do using just a dialplan? I can't see any options on the Dial command that would lead to this, am I looking in the wrong place? I looked briefly through the archive and I heard mentions of AGI, is this what must be used to accomplish this? If you want it to be detected within a call, which is what I'd assume, you'll probably be looking at the applicationmap section within features.conf. http://www.voip-info.org/wiki/view/Asterisk+config+features.conf -- _ -- 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
[asterisk-users] Music on Hold
Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
The files used to be Frederic Chopin – Polonaised Op. 40-2.raw I have since replaced the raw files with the original mp3s They are now as follows: [r...@tsunami musiconhold]# ls -l . total 13320 -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3 -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3 I also have the same issue with the default files in /var/lib/asterisk/moh . On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas da...@debsinc.com wrote: What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin – Polonaised Op. 40-2.wav?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] Music on Hold
This might be another piece of the puzzle: It would appear any application using playback functionality exits immediately. For example anything involving voicemail or playback. Phone calls work with no problem but not if asterisk must play something back. The modules are loaded however... Tsunami*CLI module show like voicemail Module Description Use Count app_voicemail.so Comedian Mail (Voicemail System) 0 I'm begining to think that the problem lies with my vendor's package. On Wed, Sep 16, 2009 at 5:30 PM, Dan Saul daniel.s...@gmail.com wrote: The files used to be Frederic Chopin – Polonaised Op. 40-2.raw I have since replaced the raw files with the original mp3s They are now as follows: [r...@tsunami musiconhold]# ls -l . total 13320 -rw-r--r-- 1 asterisk asterisk 5415926 2009-09-16 17:18 hm1.mp3 -rw-r--r-- 1 asterisk asterisk 8217974 2009-09-16 17:18 hm2.mp3 I also have the same issue with the default files in /var/lib/asterisk/moh . On Wed, Sep 16, 2009 at 5:02 PM, Danny Nicholas da...@debsinc.com wrote: What are your actual file names (/etc/asterisk/musiconhold/Frederic Chopin – Polonaised Op. 40-2.wav?) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Music on Hold That was a good shot in the dark, but sadly renaming it to something simple (and removing all non ascii in the process) does not correct this. On Wed, Sep 16, 2009 at 4:37 PM, Danny Nicholas da...@debsinc.com wrote: Just a “shot in the dark” but could MOH be choking on the “long file names”? (does it work on fred_chopin_pol_1)? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dan Saul *Sent:* Wednesday, September 16, 2009 4:18 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Music on Hold Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI moh show files Class: default File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Frédéric Chopin - Polonaises Op. 40-1 These files were generated by SoX: Channels : 1 Sample Rate: 8000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM Endian Type: little Reverse Nibbles: no Reverse Bits : no Comment: 'Processed by SoX' This prints in the asterisk console when you attempt to put someone in hold: -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rv): == Using SIP RTP CoS mark 5 -- Executing [...@phones:1] Goto(SIP/ATA-xx-L1-024b6d88, 1xx,1) in new stack -- Goto (phones,1xx,1) -- Executing [1xxx...@phones:1] MSet(SIP/ATA-xx-L1-024b6d88, oldcidnum=0) in new stack -- Executing [1xxx...@phones:2] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(name)=) in new stack -- Executing [1xxx...@phones:3] MSet(SIP/ATA-xx-L1-024b6d88, CALLERID(num)=xx) in new stack -- Executing [1xxx...@phones:4] Monitor(SIP/ATA-xx-L1-024b6d88, wav,/tmp/out 0 2009-09-17 03h 04m 51s CST xx,m) in new stack -- Executing [1xxx...@phones:5] Gosub(SIP/ATA-xx-L1-024b6d88, ExternalDial,s,1(1xx)) in new stack -- Executing [...@externaldial:1] MSet(SIP/ATA-xx-L1-024b6d88, LOCAL(num)=1xx) in new stack -- Executing [...@externaldial:2] MSet(SIP/ATA-xx-L1-024b6d88, ~~EXTEN~~=s) in new stack -- Executing [...@externaldial:3] Dial(SIP/ATA-xx-L1-024b6d88, SIP/1xxx...@link2voip-sw1,120) in new stack == Using SIP RTP CoS mark 5 -- Called 1xxx...@link2voip-sw1 -- SIP/link2voip-sw1-02477668 is making progress passing it to SIP/ATA-xx-L1-024b6d88 -- SIP/link2voip-sw1-02477668 answered SIP/ATA-xx-L1-024b6d88 -- Started music on hold, class 'default', on SIP/link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 doing dnsmgr_lookup for 'sip.ca2.link2voip.com' doing dnsmgr_lookup for 'sip.ca1.link2voip.com' == Spawn extension (ExternalDial, s, 3) exited non-zero on 'SIP/ATA-xx-L1-024b6d88' Any thoughts or ideas? If there were an error I could work on solving that, but there is none... Thanks
