[asterisk-users] Hangup Detection Problem In Turkey
Hi, Although zonedata.c contains ITU E.180 recommendations for Turkey, we are still experiencing unrecognized hangups from Turk Telekom PSTN lines when callers hangup. Turk Telekom does *not* provide supervised disconnects on analog PSTN, and the tone we receive we when caller hangs up is similar to busy, with three short beeps, followed by one long beep, which keeps repeating. We've tried busydetect, polarityswitch, etc. with no success. As it stands, asterisk using dahdi (with Digium TDM410p) does not work in Turkey. Can offer small bounty for any developer wishing to solve this issue. The technology we have used shown below; Asterisk 1.6.2.8 Libpri 1.4.11.1 Dahdi 2.3.0 Ubuntu Server 9.10 Kind Regards -- _ -- 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] Hangup Detection
Is there any way, i can detect in asterisk that which party hanged up the call either from A side or B. Both parties are using SIP protocol. I am using Asterisk 1.4.27 Shariq Khan 0333-3501125 -- _ -- 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] Hangup Detection After Originate (Asterisk Manager API)
On 24/04/2009 2:22 p.m., Saurabh Nirkhey wrote: > I have written an asterisk manager client which creates an outbound > call using Asterisk manager API's Originate action. > when the call is connected I run 3 applications on it. > 1)read a dtmf digit from user > 2)A customized application which I have written,(It plays something to user) > 3)Hangup > > If user hangs up while app 2(see above) is executing I get a 'Event Hangup' > from asterisk in my manager client . > But if app2 is over and asterisk executes Hangup (app3),It never sends > any packet to my client regarding Hangup of the call. > > I have given all permissions to manager user in manager.conf. > Can somebody help me? Maybe use the UserEvent application before calling hangup: -= Info about application 'UserEvent' =- [Synopsis] Send an arbitrary event to the manager interface [Description] UserEvent(eventname[|body]): Sends an arbitrary event to the manager interface, with an optional body representing additional arguments. The body may be specified as a | delimeted list of headers. Each additional argument will be placed on a new line in the event. The format of the event will be: Event: UserEvent UserEvent: [body] If no body is specified, only Event and UserEvent headers will be present. -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup Detection After Originate (Asterisk Manager API)
I have written an asterisk manager client which creates an outbound call using Asterisk manager API's Originate action. when the call is connected I run 3 applications on it. 1)read a dtmf digit from user 2)A customized application which I have written,(It plays something to user) 3)Hangup If user hangs up while app 2(see above) is executing I get a 'Event Hangup' from asterisk in my manager client . But if app2 is over and asterisk executes Hangup (app3),It never sends any packet to my client regarding Hangup of the call. I have given all permissions to manager user in manager.conf. Can somebody help me? Thanks & Regards === (-: Saurabh :-) === "French is the language of love,For everything else there is 'C' " "Every search begins with beginner's luck and ends with the victor being severly tested" -Paulo Coehlo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup detection and trombining
Hi All, I hate to post yet another "bloody hangup detection issue" on the list, but I have been pulling my hair out no end of late with a hangup detection issue on 1 system (have a few others out there with TDM400's and no issue but this one is causing a real headache) The scenario is - system with TDM04B, a call comes in on a analogue line, rings internally and then diverts to a mobile on a second analogue line, so we in effect have a trombone happening where a call comes in on 1 analogue and back out on another analogue. Hangup detection seems to be working most of the time, but on a regular basis does not (about once every 2 days or so). We cannot get hangup supervision / polarity reversal or any other smart way of detecting a hangup, so are using busydetect. What seems to be happening is that on trombone'd calls when both parties hangup, there is a busy tone being played on each leg of the call back down each line. Some times we seem to get lucky and the tones are played in sync and a hangup occurs, but other times the tones are out of sync with each other and are overlapping, causing a non-normal tone on the line(s) or a continuous tone rather than a 'beep beep beep' which means the card / system cannot detect a hangup via busy detect. Can anyone out there confirm if my assumptions are correct re the dual'ing or the tones and the effect it will have on hangup detection? And if correct, can anyone recommend a work around to get hangup detection working in such a scenario? Cheers, Ben ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Hangup detection with Sangoma A200 in the UK?
I'm having real problems getting my Sangoma A200 card with FXO board in to detect hangup at all. Basically if the remote end hangs up the call, Asterisk does not seem to detect a hangup. A month or so ago I was running a system with 2 x X101P cards in, this detected hangup fine. Since switching to a Sangoma A200 based system hangup is no longer detected. I've tried tweaking busydetect, busycount etc but it seems to make no difference. I'm using a system based on Trixbox, which is Asterisk 1.2.9.1 with Zaptel 1.2.6, wanpipe drivers are Beta4-2.3.4. Is there any good way of debugging this that I have missed? thanks Mike UK ___ --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
[Asterisk-Users] Hangup Detection
Is it possible to patch the zaptel drivers (or whatever appropriate files) to use DTMF tone "D" for hangup detection? I have a Toshiba PBX which does not provide CPC by any means other than congestion or D tone. Thanks, Aaron Picht ___ --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
Re: [Asterisk-Users] Hangup Detection (revisited)
Darrick Hartman wrote: A little background. I'm integrating asterisk as the voicemail service for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2 is used to connect an analog device (such as a voice modem) to the pbx. In the past we've used vgetty and a voice modem with varying degrees of success. If you haven't yet, I'd turn on busydetect in zapata.conf. Can't hurt and might (although unlikely) work (I had to turn it on to make it work on my system). Switching to loop-start might be worth a try, too. For a while my VM * system wasn't doing disconnect detection, and it was OK. I had trouble with the single-port cheapo cards off eBay with the silence thresholds, but using a TDM400P card fixed that for me. Also make sure all you menus will time out and hang up. You could try posting to the Nortel list: http://www.tgrace.com/mailman/listinfo/nortel-list They've been very helpful and kind to me. Phil ___ --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
[Asterisk-Users] Hangup Detection (revisited)
A little background. I'm integrating asterisk as the voicemail service for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2 is used to connect an analog device (such as a voice modem) to the pbx. In the past we've used vgetty and a voice modem with varying degrees of success. The problem is the ATA-2 does not provide disconnect supervision. When the outside caller hangs up, the line just goes silent. No busy signal. If the inside analog device hangs up the call, the ATA-2 then provides a dial tone. Right now, we've been using silence detection in voicemail to determine when the call ends. This works most of the time. On longer calls (say more than 5 minutes), sometimes the silence detection does not function and the call continues (with a bunch of recorded silence) until the maximum voicemail length limit is reached. Obviously, busy-detect is not going to work. The current silence threshold is set at the default 128. The FXO module is configured with Kewl Start signalling. Would I get better results (or any different results) if I switched to Loop Start? Any other ideas? Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ --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
Re: [Asterisk-Users] hangup detection
On Tue, 10 Jan 2006, [EMAIL PROTECTED] wrote: > Thanks for your suggestion Steve. > I have done as you advised and set busypattern=300,200 to match the sample > I recorded. > This hasn't worked though, asterisk doesn't seem to detect the busy signal. > Does asterisk require a the signal to be in a certain power range? The > signal I get > is very quiet. > Thanks for your help > Regards > Jonathan Yeah - it needs to be reasonably loud to be detected. Too bad. Steve ___ --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
FW: Re: [Asterisk-Users] hangup detection
Thanks for your suggestion Steve. I have done as you advised and set busypattern=300,200 to match the sample I recorded.This hasn't worked though, asterisk doesn't seem to detect the busy signal.Does asterisk require a the signal to be in a certain power range? The signal I getis very quiet.Thanks for your helpRegardsJonathan On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote: > Hi everybody! > > Jonathan wrote: > > > > Hi, > > > > I'm using a td400p card with an FXO port and asterisk 1.2.1 in South > > Korea and asterisk isn't detecting when PSTN callers hangup. > > I've gone through all the settings related to hangup detection and none > > work. I've tried: > > hanguponpolarityswitch=yes > > callprogress=yes > > busydetect=yes > > busycount=6 > I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in > Colombia and tried with a lof of loadzone= > > > > Debug doesn't show reverse polarity events so I'm pretty stuck. > > > > I've got zaptel configured with a loadzone of US and kewlstart signialling. > > > > Has anybody had success with these cards/asterisk in South Korea? > ?Or in the world? > > We implemented a busypattern= option for the zapata.conf that might help you. Test like so: Dial into your Asterisk system via the FXO port to an extension on your box. Now hang up from the outside. Listen to the call on the internal extension. If you hear a regular beep-beep tone of some sort, busypattern= might help you. You need to time exactly the length of the beep and the length of the silence. (To get it nice and accurate, record it, then load into Audacity and measure). Say it comes out at 750 msec of beep, 500 msec of silence. Then adjust your zapata.conf like so: busydetect=yes callprogress=no busypattern=750,500 busycount=4 Regards, Steve Davies _ ___ --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
Re: [Asterisk-Users] hangup detection
On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote: > Hi everybody! > > Jonathan wrote: > > > > Hi, > > > > I'm using a td400p card with an FXO port and asterisk 1.2.1 in South > > Korea and asterisk isn't detecting when PSTN callers hangup. > > I've gone through all the settings related to hangup detection and none > > work. I've tried: > > hanguponpolarityswitch=yes > > callprogress=yes > > busydetect=yes > > busycount=6 > I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in > Colombia and tried with a lof of loadzone= > > > > Debug doesn't show reverse polarity events so I'm pretty stuck. > > > > I've got zaptel configured with a loadzone of US and kewlstart signialling. > > > > Has anybody had success with these cards/asterisk in South Korea? > ?Or in the world? > > We implemented a busypattern= option for the zapata.conf that might help you. Test like so: Dial into your Asterisk system via the FXO port to an extension on your box. Now hang up from the outside. Listen to the call on the internal extension. If you hear a regular beep-beep tone of some sort, busypattern= might help you. You need to time exactly the length of the beep and the length of the silence. (To get it nice and accurate, record it, then load into Audacity and measure). Say it comes out at 750 msec of beep, 500 msec of silence. Then adjust your zapata.conf like so: busydetect=yes callprogress=no busypattern=750,500 busycount=4 Regards, Steve Davies ___ --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
Re: [Asterisk-Users] hangup detection
Hi everybody! Jonathan wrote: > > Hi, > > I'm using a td400p card with an FXO port and asterisk 1.2.1 in South > Korea and asterisk isn't detecting when PSTN callers hangup. > I've gone through all the settings related to hangup detection and none > work. I've tried: > hanguponpolarityswitch=yes > callprogress=yes > busydetect=yes > busycount=6 I'm using asterisk/zaptel 1.0.10 and have the same situation. I'm in Colombia and tried with a lof of loadzone= > > Debug doesn't show reverse polarity events so I'm pretty stuck. > > I've got zaptel configured with a loadzone of US and kewlstart signialling. > > Has anybody had success with these cards/asterisk in South Korea? ¿Or in the world? > > Thanks > JC > > > > > > ___ > --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 -- Diego Andrés Asenjo González Universidad del Cauca Ingeniero en Electrónica y Telecomunicaciones signature.asc Description: OpenPGP digital signature ___ --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
[Asterisk-Users] hangup detection
Hi, I'm using a td400p card with an FXO port and asterisk 1.2.1 in South Korea and asterisk isn't detecting when PSTN callers hangup. I've gone through all the settings related to hangup detection and none work. I've tried: hanguponpolarityswitch=yes callprogress=yes busydetect=yes busycount=6 Debug doesn't show reverse polarity events so I'm pretty stuck. I've got zaptel configured with a loadzone of US and kewlstart signialling. Has anybody had success with these cards/asterisk in South Korea? Thanks JC ___ --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
Re: [Asterisk-Users] Hangup detection - TDM400P
Here in Spain we had that problem since the hangup here is done by changing line polarity. It is solved by aplying this patch: http://www.maxosystem.net/asterisk/asterisk-stable-polarity-v5.diff $ cd /usr/src/asterisk/channels $ patch chan_zap.c < /your/route/here/asterisk-stable-polarity-v5.diffand in zapata.conf :answeronpolarityswitch=yes hanguponpolarityswitch=yesHope it helps ;)- Original Message - From: "Marco Supino" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, November 17, 2005 5:20 PM Subject: Re: [Asterisk-Users] Hangup detection - TDM400P Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino <[EMAIL PROTECTED]> wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have "disconnect supervision", and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Hangup detection - TDM400P
Yes, didnt change anything Marco. Angelito Manansala wrote: hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino <[EMAIL PROTECTED]> wrote: Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have "disconnect supervision", and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] Hangup detection - TDM400P
hmmm di you try this ;hanguponpolarityswitch=yes Cheerz! On 11/17/05, Marco Supino <[EMAIL PROTECTED]> wrote: > Hi, > > I have a long delay when detecting hangups on the TDM400P card, with 4 > FXO ports, > > When an incoming call dial's in, when hanging up, the asterisk will > detect the hangup only after 10 seconds, i searched around, and found > many similar problems, but no solution, i tried some options in > zapate.conf , but nothing helped, any solution ? > > the lines are coming from SBC in San Fransisco, i asked them if i have > "disconnect supervision", and they said i do have it. > > Marco. > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hangup detection - TDM400P
Hi, I have a long delay when detecting hangups on the TDM400P card, with 4 FXO ports, When an incoming call dial's in, when hanging up, the asterisk will detect the hangup only after 10 seconds, i searched around, and found many similar problems, but no solution, i tried some options in zapate.conf , but nothing helped, any solution ? the lines are coming from SBC in San Fransisco, i asked them if i have "disconnect supervision", and they said i do have it. Marco. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Hangup Detection with busydetect
My telco doesn't provide Disconnect Supervision or Polarity Change. So I figured I have to detect hangups with busydetect=yes in zapata.conf. I tested it. When the telco sends a busy tone * detects it and hangsup. So far so good. The problem is the telco doesn't always send a busy after remote hangup. Most of the time it sends a congestion tone. I am guessing these tones from what I read on indications.conf. "diitdiitdiit" for busy "diit diit diit diit" for congestion busy = 450/500,0/500 congestion = 450/200,0/200,450/200,0/200,450/200,0/200,450/600,0/200 Looks like I have the correct setting for my country in indications.conf, verified it with ITU tones document. So at this point I figure I need to somehow make * detect both busy and congestion as same and hangup. I tried different BUSYDETECT algorithms, poked around at source code. Couldn't figure it out. Just to test what happens, I tried to change the tones for busy and replace it with the tones for congestion in indications.conf. To my surprise * continued to detect the old busy tone correctly and ignored the new tones I put in. I did the same in zaptel/zonedata.c and still * continues to detect the old busy tone correctly and ignores the new one I put in. So at this point I am totally confused. I don't even know where * gets the information about the tones. I am using CVS-HEAD as of today. Thank you -- Mehmet ___ 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
Re: [Asterisk-Users] Hangup detection on Panasonic KXTD816
Hilton Williams wrote: Hi I have a Digium TDM400 card with 4 FXO modules connected to the extension ports on a Panasonic KXTD816. I'm using [EMAIL PROTECTED] v1.0, which has Asterisk 1.07. There's a problem that Asterisk doesn't detect when the line is disconnected on the Panasonic. The Panasonic doesn't provide polarity reversal or current drop or anything like that to indicate hangup. It just plays the dial tone again. Correct. When I have to interface with a PBX I use FXS ports on Asterisk connected to the FXO/CO ports of the PBX. This seems to (mostly) work well, since PBXs tend to be MUCH better at figureing out that a line is disconnected than Asterisk is. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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] Hangup detection on Panasonic KXTD816
Hi I have a Digium TDM400 card with 4 FXO modules connected to the extension ports on a Panasonic KXTD816. I'm using [EMAIL PROTECTED] v1.0, which has Asterisk 1.07. There's a problem that Asterisk doesn't detect when the line is disconnected on the Panasonic. The Panasonic doesn't provide polarity reversal or current drop or anything like that to indicate hangup. It just plays the dial tone again. I've tried several different things, thinking that the settings were not correct for South Africa. I've tried adding busydetect=yes and busycount=4, since that worked for our analogue Telecom lines (they work the same way as the Panasonic). At the moment, I'm thinking I need to change the busy tone in indications.conf. Currently, it is set to 400/500,0/500, which are the settings for Alcatel switches in the Western Cape, South Africa. The PABX seems to need 400/250,0/250, but that doesn't work. Has anybody tried this before? It's probably not specific to South Africa, but Panasonic KXTD PABXs. Regards Hilton Datatex Dynamics CC Web site http://www.datatex.co.za/ Email to [EMAIL PROTECTED] Tel +27215924033Fax +27215924077 The use of the Datatex e-mail facility is not permitted for the distribution of chain letters or offensive email of any nature whatsoever. Datatex hereby distances itself from and accepts no liability in respect of the unauthorised use of its e-mail facility or the sending of e-mail communications for other than strictly business purposes. Datatex furthermore disclaims liability for any unauthorised instruction for which permission was not granted. Any recipient of an unacceptable communication, a chain letter or offensive material of any nature is requested to report it to [EMAIL PROTECTED]. ___ 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
Re: [Asterisk-Users] Hangup detection with TDM400 in UK
On Tue, Feb 08, 2005 at 04:03:50PM +0200, Doug Reid - Stormcorp wrote: > Hi > > Try going into "vi /etc/profile" insert the lines in brackets. > > > > USER="`id -un`" > LOGNAME=$USER Generally LOGNAME is set by login, sshd or whatever program you login with. If this is linux you generally shouldn't change those parts of /etc/profile unless you have a "strange" setup and you know what you're doing and you're absolutely sure you know what you're doing. > MAIL="/var/spool/mail/$USER" Correct with the default settings. However many programs assume that (or /var/mail/$USER ) > MONITOR_EXEC=/usr/bin/soxmix > VPB_TONE=BUSY,P,400,100,500(insert the following line) > > > HOSTNAME=`/bin/hostname` > HISTSIZE=1000 > > if [ -z "$INPUTRC" -a ! -f "$HOME/.inputrc" ]; then > INPUTRC=/etc/inputrc > fi Handy, but only for an interactive shell. Don't add it to your /etc/profile unless you know where to place it. and it may be a bit distro-specific. > > export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC MONITOR_EXEC > VPB_TONE (and insert here VPB_TONE) > > > == > > The 400 100 and 500 are related to your country use indications file for > info > on what those values should be. > > Regards > Doug Most Linux distros have /etc/profile.d where you can add your custom .sh scriptlets to add some vaiables without stepping over the default /etc/profile so you don't have you worry about upgrades of the distro's packages. In my Debian system there is no such thing. However I see /etc/environment with has lines of 'var=value', and the pam module pam_env.so loads it at login time. I don't know if this is used with other distros. -- Tzafrir Cohen | New signature for new address and | VIM is http://tzafrir.org.il | new homepage | a Mutt's [EMAIL PROTECTED] || best ICQ# 16849755 | Space reserved for other protocols | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangup detection with TDM400 in UK
Hi Try going into "vi /etc/profile" insert the lines in brackets. USER="`id -un`" LOGNAME=$USER MAIL="/var/spool/mail/$USER" MONITOR_EXEC=/usr/bin/soxmix VPB_TONE=BUSY,P,400,100,500(insert the following line) HOSTNAME=`/bin/hostname` HISTSIZE=1000 if [ -z "$INPUTRC" -a ! -f "$HOME/.inputrc" ]; then INPUTRC=/etc/inputrc fi export PATH USER LOGNAME MAIL HOSTNAME HISTSIZE INPUTRC MONITOR_EXEC VPB_TONE (and insert here VPB_TONE) == The 400 100 and 500 are related to your country use indications file for info on what those values should be. Regards Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Patrick Lidstone (Personal E-mail) Sent: Wednesday, February 02, 2005 2:03 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Hangup detection with TDM400 in UK When a caller hangs up (e.g. after leaving a voicemail), my British Telecom exchange sends a continuous tone for about 15s and then silence. I can't get asterisk to recognise this tone as a hangup indication. I have tried indications.conf with both country=uk and country=us. My zapata.conf has busydetect=yes, callprogress=yes and I've tried setting busycount from 1 through 7 I am using kewlstart signalling on the FXO module. Any suggestions gratefully received - I really don't want to resort to using an absolute timeout. Thanks Patrick ___ 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
[Asterisk-Users] Hangup detection with TDM400 in UK
When a caller hangs up (e.g. after leaving a voicemail), my British Telecom exchange sends a continuous tone for about 15s and then silence. I can't get asterisk to recognise this tone as a hangup indication. I have tried indications.conf with both country=uk and country=us. My zapata.conf has busydetect=yes, callprogress=yes and I've tried setting busycount from 1 through 7 I am using kewlstart signalling on the FXO module. Any suggestions gratefully received - I really don't want to resort to using an absolute timeout. Thanks Patrick ___ 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
Re: [Asterisk-Users] Hangup Detection
Ali Mughrabi wrote: Hi , I need to execute a query when a user hangs up the agi application , I’ve tried monitoring some return values of AGI commands Still doesn’t work . Any ideas ? Thanx Ali Mughrabi You will need to put another agi with you cleanup script onto the 'h' extension.. If you generate a call with agi the agi will continue and complete after the call is generated so is not running when the call is terminated.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup Detection
Hi , I need to execute a query when a user hangs up the agi application , Ive tried monitoring some return values of AGI commands Still doesnt work . Any ideas ? Thanx Ali Mughrabi MSN 8 helps ELIMINATE E-MAIL VIRUSES. Get 2 months FREE*. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection failed
> > Use something like the following in voicemail.conf > > ; How many seconds of silence before we end the recording > > maxsilence=10 > > ; Silence threshold (what we consider silence, the lower, the more sensitive) > > silencethreshold=128 > > > > Rich > > Ah, great. Thanks! Do you know how to find out what the current settings > are? (I guess it must have been too sensitive). Look for the entries above in your voicemail.conf file. If there are no entries, then I'm not sure what the default values are but it would appear from your original post the default is to record forever. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection failed
> Use something like the following in voicemail.conf > ; How many seconds of silence before we end the recording > maxsilence=10 > ; Silence threshold (what we consider silence, the lower, the more sensitive) > silencethreshold=128 > > Rich Ah, great. Thanks! Do you know how to find out what the current settings are? (I guess it must have been too sensitive). - Kim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangup detection failed
> We have a system that recorded voicemail for about an hour after the caller
> hungup. I'm going to put a timeout on it but is there anything to look for
> that can help prevent this? The system is running on a telenet line in
> Belgium. The answer dialplan I used was:
>
> [macro-stddial]
> exten => s,1,Answer
> exten => s,2,Playback(transfer)
> exten => s,3,Dial(${ARG2},60)
> exten => s,4,Voicemail(u${ARG1})
> exten => s,5,Playback(tt-monkeysintro)
> exten => s,6,Playback(vm-goodbye)
> exten => s,7,Hangup
> exten => s,104,Voicemail(b${ARG1})
Use something like the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
Rich
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[Asterisk-Users] Hangup detection failed
Hi,
We have a system that recorded voicemail for about an hour after the caller
hungup. I'm going to put a timeout on it but is there anything to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten => s,1,Answer
exten => s,2,Playback(transfer)
exten => s,3,Dial(${ARG2},60)
exten => s,4,Voicemail(u${ARG1})
exten => s,5,Playback(tt-monkeysintro)
exten => s,6,Playback(vm-goodbye)
exten => s,7,Hangup
exten => s,104,Voicemail(b${ARG1})
- Kim Hendrikse
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[Asterisk-Users] Hangup detection issue
Hello all, I just went into production on a system. 6 incomming lines 24 phones (* setting in front of an old key system). The six lines are standard analog POTS lines through a service called Centrex/Plexar. I am using an CAC Adit 600 channel bank with 1 8 port fxs and 1 8 port fxo cards. I patch the lines into the * server with a T1 card and T1 cross over cable to the CB. The system is working really well except that it takes about 8 seconds for * to acknowledge a hang up. Each person answering the phone reported a lot of instances of the phone ringing but when answered no one being there. Additionally, we had 30 voice mail messages today all but three of which were empty messages. I believe this has something to do with a feature called call supervision. How do I make * and or the Adit 600 detect the hangups properly? -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Okay, I'm an idiot. The tones are picked up just fine by asterisk with no changes. It helps if you understand the syntax of zapata.conf. I thought busydetect=yes just had to be under the context line. I didn't realize how the "channels=" is actually the delimiter that includes the stuff above it (I had busydetect below that line). I should add that I find the asterisk config files to be very whacky in general. On Jan 2, 2004, at 12:34 PM, Martin Pycko wrote: If the on/off times are diffrent you need to edit Makefile and uncomment BUSYDETECT_TONES_ONLY flag or something like that ... and then you can change the MAX/MIN values in dsp.c too. That should help you with busycount=10 and busydetect=yes regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: Here's a recording: http://www.seanadams.com/hangup_tones.aif (sorry - recorded from speakerphone - skip to the end) The following numbers are not real precise, I just got this from visually looking at the spectrum on my computer: The tones appear to consist of 2600, 2440, 2000, and 1400 Hz. The timing is 120ms on, 80ms off. I'll take a look at dsp.c and see if I can make it work. Thanks for the pointers. On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
If the on/off times are diffrent you need to edit Makefile and uncomment BUSYDETECT_TONES_ONLY flag or something like that ... and then you can change the MAX/MIN values in dsp.c too. That should help you with busycount=10 and busydetect=yes regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: > > Here's a recording: > > http://www.seanadams.com/hangup_tones.aif > > (sorry - recorded from speakerphone - skip to the end) > > The following numbers are not real precise, I just got this from > visually looking at the spectrum on my computer: > > The tones appear to consist of 2600, 2440, 2000, and 1400 Hz. > > The timing is 120ms on, 80ms off. > > I'll take a look at dsp.c and see if I can make it work. Thanks for the > pointers. > > > > On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: > > > busydetect should help you. Set busycount=10 busydetect=yes in > > zapata.conf > > and measure the length of the tone .. should be equal the pause too. > > > > Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like > > this: your result - 100, your result + 100 [ms] > > > > regards > > Martin > > > > On Fri, 2 Jan 2004, Sean Adams wrote: > > > >> > >> So I made the mistake of buying a Carrier Access channel bank without > >> noticing the page on the wiki about the fact that they don't support > >> disconnect supervision (bastards!). However, apart from that, I do > >> have > >> it working fine for incoming calls. > >> > >> Is there some trick to get asterisk to detect the hangup tones from > >> SBC? I've tried busydetect and callprogress as suggested, but neither > >> seems to work. The tone is not a busy tone, but that ear-piercing > >> high > >> pitched buzzer. It goes "if you'd like to make a call, please hang up > >> and try again. If you need help, hang up and then dial your operator. > >> BEEP BEEP BEEP etc." > >> > >> I am set up here with recording gear and spectrum analyzer software, > >> so > >> I can identify the tones and timing if necessary. However I'm not sure > >> how to make asterisk detect the tones, or if this work has already > >> been > >> done. Anyone know? > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
On Fri, 2004-01-02 at 13:02, Sean Adams wrote: > > > > Are the tones increasing in pitch? > > No, the beeps are the same pitch - sounds like it was deliberately > designed to be a loud and awful sounding as possible through an > off-hook phone, to get your attention to go hang it up. My ears tell me > it's roughly 250ms on, 250ms off and so on. Okay. > Taking my first peek at the code now... Always a good thing. > > BTW, which CAC channel bank did you buy? The ADIT 600 should do > > disconnect supervision, and I thought the AB1 did too. > > It's the AB1 with 8 fxo, 16xfs. Here's the page I was talking about: > > http://www.voip-info.org/wiki-Asterisk+hardware > > Also, others have reported this problem but I can't find a resolution: > > http://www.mail-archive.com/[EMAIL PROTECTED]/msg18626.html > > > Are you also sure > > you have that on your line so as to be detected? Your other option > > might > > be to switch to groundstart lines which detect hangup much easier. May > > be difficult to get unless you are a business though. > > I just have regular business lines without any special provisioning. I > don't understand why a $20 answering machine can do this but an > expensive channel bank can't. :( The difference is acceptable failure. If your $20 answering machine fails by hanging up early, they only one really annoyed is the person leaving a message and they will think they hit a record length limit unless it was pretty short. If you are placing a call though the machine and it thought the other side hung up and so it disconnected your conversation, you would consider that unacceptable. The other part is that disconnect supervision is something that basically breaks the loop long enough, or reverse polarity for a moment to let the other side disconnect. Think about how a relay would work, reverse polarity or disconnect battery and it will disconnect the points. Now days, that type of technology is rarely used, and therefore not implemented unless asked for. It is highly probably that you don't have disconnect supervision on your phone line. You should be able to hook up your test equipment and see it. I think it has been discussed here before about using a phone that takes power from the line to light up, if it blinks when the other side hangs up, you have disconnect supervision. Otherwise, it will always be a problem detecting hangup without waiting for those tones and matching on them. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Not having any luck with just tweaking those values. I'm a bit confused still as to how the different busy detection choices are supposed to work - I've uncommented a few of the #if 0 to see if it's doing anything, and I can't see any indiciation that it is. Don't the specific off-hook tones need to be in dsp.c, or is it intended that asterisk should match the signal just by the timing? Here's some information I found which confirms the tones I measured: http://www.hackfaq.org/telephony-27.shtml -- Receiver Off-Hook Tone This tone is used to cause off-hook customers to replace the receiver on-hook on a permanent signal call and to signal a non-PBX off-hook line when ringing key is operated by a switchboard operator. Receiver Off-Hook Tone is 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz at 0 dBm0/frequency on and off every .1 second. On some older space division switching systems Receiver Off-Hook was 1400 Hz, 2060 Hz, 2450 Hz and 2600 Hz at +5 VU on and off every .1 second. On a No. 5 ESS this continues for 30 seconds. On a No. 2/2B ESS this continues for 40 seconds. On some other AT&T switches there are two iterations of 50 seconds each. - On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Here's a recording: http://www.seanadams.com/hangup_tones.aif (sorry - recorded from speakerphone - skip to the end) The following numbers are not real precise, I just got this from visually looking at the spectrum on my computer: The tones appear to consist of 2600, 2440, 2000, and 1400 Hz. The timing is 120ms on, 80ms off. I'll take a look at dsp.c and see if I can make it work. Thanks for the pointers. On Jan 2, 2004, at 10:46 AM, Martin Pycko wrote: busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
Are the tones increasing in pitch? No, the beeps are the same pitch - sounds like it was deliberately designed to be a loud and awful sounding as possible through an off-hook phone, to get your attention to go hang it up. My ears tell me it's roughly 250ms on, 250ms off and so on. Are they the Special Information Tones (SIT) that are also on the message when you dial a number that has been disconnected? No, not like that at all. I'll make a recording. If so, then they are defined somewhere in the code, at least as part of app_zapateller since that is how it tries to get rid of telemarketers. You could then see about adding that to the dsp routines to detect the SIT tones and determine what to do at that time. Taking my first peek at the code now... BTW, which CAC channel bank did you buy? The ADIT 600 should do disconnect supervision, and I thought the AB1 did too. It's the AB1 with 8 fxo, 16xfs. Here's the page I was talking about: http://www.voip-info.org/wiki-Asterisk+hardware Also, others have reported this problem but I can't find a resolution: http://www.mail-archive.com/[EMAIL PROTECTED]/ msg18626.html Are you also sure you have that on your line so as to be detected? Your other option might be to switch to groundstart lines which detect hangup much easier. May be difficult to get unless you are a business though. I just have regular business lines without any special provisioning. I don't understand why a $20 answering machine can do this but an expensive channel bank can't. :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf and measure the length of the tone .. should be equal the pause too. Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like this: your result - 100, your result + 100 [ms] regards Martin On Fri, 2 Jan 2004, Sean Adams wrote: > > So I made the mistake of buying a Carrier Access channel bank without > noticing the page on the wiki about the fact that they don't support > disconnect supervision (bastards!). However, apart from that, I do have > it working fine for incoming calls. > > Is there some trick to get asterisk to detect the hangup tones from > SBC? I've tried busydetect and callprogress as suggested, but neither > seems to work. The tone is not a busy tone, but that ear-piercing high > pitched buzzer. It goes "if you'd like to make a call, please hang up > and try again. If you need help, hang up and then dial your operator. > BEEP BEEP BEEP etc." > > I am set up here with recording gear and spectrum analyzer software, so > I can identify the tones and timing if necessary. However I'm not sure > how to make asterisk detect the tones, or if this work has already been > done. Anyone know? > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hangup detection
On Fri, 2004-01-02 at 12:25, Sean Adams wrote: > So I made the mistake of buying a Carrier Access channel bank without > noticing the page on the wiki about the fact that they don't support > disconnect supervision (bastards!). However, apart from that, I do have > it working fine for incoming calls. > > Is there some trick to get asterisk to detect the hangup tones from > SBC? I've tried busydetect and callprogress as suggested, but neither > seems to work. The tone is not a busy tone, but that ear-piercing high > pitched buzzer. It goes "if you'd like to make a call, please hang up > and try again. If you need help, hang up and then dial your operator. > BEEP BEEP BEEP etc." > > I am set up here with recording gear and spectrum analyzer software, so > I can identify the tones and timing if necessary. However I'm not sure > how to make asterisk detect the tones, or if this work has already been > done. Anyone know? Are the tones increasing in pitch? Are they the Special Information Tones (SIT) that are also on the message when you dial a number that has been disconnected? If so, then they are defined somewhere in the code, at least as part of app_zapateller since that is how it tries to get rid of telemarketers. You could then see about adding that to the dsp routines to detect the SIT tones and determine what to do at that time. BTW, which CAC channel bank did you buy? The ADIT 600 should do disconnect supervision, and I thought the AB1 did too. Are you also sure you have that on your line so as to be detected? Your other option might be to switch to groundstart lines which detect hangup much easier. May be difficult to get unless you are a business though. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hangup detection
So I made the mistake of buying a Carrier Access channel bank without noticing the page on the wiki about the fact that they don't support disconnect supervision (bastards!). However, apart from that, I do have it working fine for incoming calls. Is there some trick to get asterisk to detect the hangup tones from SBC? I've tried busydetect and callprogress as suggested, but neither seems to work. The tone is not a busy tone, but that ear-piercing high pitched buzzer. It goes "if you'd like to make a call, please hang up and try again. If you need help, hang up and then dial your operator. BEEP BEEP BEEP etc." I am set up here with recording gear and spectrum analyzer software, so I can identify the tones and timing if necessary. However I'm not sure how to make asterisk detect the tones, or if this work has already been done. Anyone know? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hangup Detection and BUSYDETECT_MARTIN
Hello, I've got the following configuration: 2 X101Ps Asterisk built with BUSYDETECT_MARTIN busydetect=yes busycount=10 callprogress=yes signalling = fxs_ks With this setup, the best I can do is get voicemail with 17 to 19 seconds of silence tacked on at the end. Ideally, I'd like at most 2-5 seconds. Has anyone had any success with this? It seems that hangups are indeed detected, since if I fiddle with the above configuration much then the silence at the end of the message increases dramatically. Is there a soft timeout for silence detection somewhere? Where should I be looking in the code to understand the silence detection stuff? dsp.c and app_voicemail.c? Thanks for any help, Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
