similar to: Patch to avoid sporadic hangup during normal conversation

Hi All; Why busydetect=yes caused this autmatic hangup happens without any reson (while responding to entering the digits in the IVR) I do not know! And what is the solution I do not know. I used busydetect=yes and busycount=5 in zapata.conf to help me in hangup when detect the busy tone, but I faced the following problem: While I am calling to the Asterisk PBX, and during the response for the

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

Good day , I have configured TDM410P (asterisk 1.6.x) on Cent OS 5, but dahdi take some time to hangup the call when playing the IVR..(it will send the hangup signal after finishing the IVR promt..) is there any specific setting to avoid such incidents ? iam using busycount as 3, signalling=fxs_ks ;toneduration=100 callwaiting=yes threewaycalling=yes callreturn=yes

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

install asterisk with x100p clone; the problem is that call me and hangup but the interface zap not detect the hangup and the line open. the error is : The 'reload' command is deprecated and will be removed in a future release. Please use 'module reload' instead. -- Reloading module 'chan_zap.so' (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf':

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

Hi ! I have some issues, i don't know exactly if it's a busy detection issue. When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk hangs up too. I have this parameters on zapata.conf: busydetect=no

Hi, Does some one experience the Sangoma A20X-ec series card that cant detect the hangup tone? I got * server 1.2.5 and running on Centos 4.3, I hock up to two PSTN lines and each time some one call in and my phone delay 1-2 sec (this is Asterisk delay nothing to do with Sangoma) and it rings on my phone, however, end on the day I got not less that 10 empty messages. I found out that Sangoma FXO

Hi, Probably a stupid syntax problem.. but can't seem to make these files work for more than the first T1 (ok, if I comment out the 2nd, 3rd, and 4th span). Can someone proofread this for me? Thanks! ====================== zaptel.conf span=1,0,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,esf,b8zs bchan=25-47 dchan=48 span=3,0,0,esf,b8zs bchan=49-71 dchan=72 span=4,0,0,esf,b8zs bchan=73-95

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

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

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation?

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,

Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can

Hello, I am facing hangup delay problem using FXO (X100P) card with the following scenario: PSTN Phone--------->BTTB PSTN Gateway------->Asterisk BOX-FXO-------->SIP Endpoint When anyone make call to my Asterisk-BOX FXO, they gets IVR to press their desired SIP extension number. If calling party disconnect the his/her call before pick up the call by the called party, then called

Hi There, Thanks for all your suggestions. I have now compiled asterisk from cvs running on FD4. I have performed all the suggested configurations: > busydetect=yes ;changed 17.03.04 from no > busycount=7 ; added as above > for me the distro asterisk package didnt hang up properly on busy > > signal. I > needed to download the source and uncomment BUSYDETECT_MARTIN

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 All, This isn't really a question, but it's an issue I experienced that was driving me crazy for a few days, so I thought it might be good for the archives. Basically what was happening was everytime a particular customer called (long distance), the line would disconnect immediately after answering. I thought it might have been the phone, so I swapped the phone with another - still

Asterisk Users, I am running Asterisk-1.4.11 on a Debian "Etch" system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions,

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)