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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 On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Hello people, this is my first mail to this list. I'm new to asterisk and trying to set up an IVR. So far my dialplan works nice connecting with softphones, but I'm having problems to detect hangups on the analog line. Here are the details: -Clone pc with Ubuntu server 10.04 64 bits -Asterisk 1.6 -Openvox A400P, lspci -v says: 01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX

Hi, I am using TDM400 card with 3 fxs and 1 fxo. I am struggling to detect hangup tone or disconnect supervision tone from my CO. I attached the recorded wav file which contains my telco's disconnect supervision. I am using , asterisk-1.4.33.1 dahdi-linux-complete-2.3.0.1+ 2.3.0 OS => Debian-lenny 5 users.conf ------------- [trunk_1] trunkname = pstn ; GUI

When I call my Asterisk-server from my cell phone on one of the PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card, and in the dialplan the end of a context is reached and Asterisk needs to execute the Hangup()-command, I notice the following : - Asterisk tells me that the conversation was hung up (the log files tell me the command was executed) - On my cell phone I hear

When I enable callprogress I get this error message... (when I call, it will ring forever but asterisk is acting as if it DID pick up the line ... but it never did... ) I'm beginning to think that asterisk can't hangup the line when in voice mail and this, whatever action I'll take... Actually, callprogress doesn't seem to work at all... (in my case) Mar 24 23:37:28

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1

Hello, I've commited the new busydetect routine to CVS. You need to cvs update asterisk of course and then choose it in asterisk/Makefile and recompile asterisk. All you X100P users that had the problems with false hangups or the card not being able to detect the busy tone please check that. In the asterisk/Makefile you need to find a line BUSYDETECT = and uncomment what you want/ comment

Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn?t work and i was told that 1.6 can?t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.20.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.5.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

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

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

On a test fax: -- Executing [s at incoming-fax:1] Set("DAHDI/4-1", "FAXFILE=/var/spool/asterisk/fax/20110415_1825") in new stack -- Executing [s at incoming-fax:2] Answer("DAHDI/4-1", "") in new stack -- Executing [s at incoming-fax:3] ReceiveFAX("DAHDI/4-1", "/var/spool/asterisk/fax/20110415_1825.tif") in new stack

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

/*! \brief Wait up to 16 seconds for first digit (FXO logic) */ static int firstdigittimeout =3D 16000; /*! \brief How long to wait for following digits (FXO logic) */ static int gendigittimeout =3D 8000; /*! \brief How long to wait for an extra digit, if there is an ambiguous ma= tch */ static int matchdigittimeout =3D 3000; -----Original Message----- From: asterisk-users-bounces at

Hi, I have a SPA-3000. My telco sends no disconnect indication whatsoever, so I am having trouble making the PSTN line detect disconnect. What I want it to do is detect the very loud off-hook tone that is generated about 20 seconds after disconnect. It's not perfect, but it's a lot better than silence detection. I noticed that the zaptel drivers can detect this with busydetect, but