similar to: Manager interface, again

Hi there, I have a new asterisk working in New Zeland and everything is working well except when an incoming call to the PSTN hangs up, asterisk wont hang up the zap trunk (X100P). I have found this information: http://bugs.digium.com/bug_view_page.php?bug_id=0001474 Which discusses my problem and i have made sure that i have the latest info in the indications.conf as follows: [general]

Hi all How do one "record" the tone that a pbx gives for hangup, so that one can use that in indications.conf? thanks --------------------------------- Do you Yahoo!? Yahoo! Search presents - Jib Jab's 'Second Term' -------------- next part -------------- An HTML attachment was scrubbed... URL:

Any body used patch, http://bugs.digium.com/bug_view_page.php?bug_id=0001719 to get the callerid for BT Line. I applied the patch successfully but could not get it to work. Any help. Here are the logs: -- Starting simple switch on 'Zap/1-1' Jun 17 18:22:31 NOTICE[426000]: chan_zap.c:4811 ss_thread: Got event 2 (Ring/Answered)... Jun 17 18:22:34 NOTICE[426000]: chan_zap.c:4811

Hi I know this gets covered fairly regularly, but I've had a search through the archives and can't find anything dealing with this specifically - apologies if I've missed it though. I've got a TDM11B with the fxo port plugged into a standard UK BT PSTN line, loading wcfxs with OPERMODE=UK. All's working well, except if I get an incoming call through my bt line, and the

Folks, For the last few days I've been trying to experiment with a Panasonic PBX and an X100P but have run into quite a few problems which I am not sure if they can be solved with this type of card (how about TDM01B?) 1) I wanted to use *'s IVR capabilities, so I routed the calls to the extension where the x100p was connected to. Asterisk should answer the call, playback a message,

I'm attempting to use callprogress in my system, and I'm having trouble. Callprogress always can tell if the line is busy or ringing, but when the line is answered, the call does not get bridged. Messages showing that "line is ringing" stop in the console and if the called party hangs up, asterisk reports the line is busy. Are there any settings that I could use to help with

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hi Guys, I am getting a complain that call on analogue lines (Sangoam A400D) drops all of a sudden. Here is what I see in logs: [Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy, avgtone: 75, avgsilence 135 [Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing [h at macro-dialout-trunk:1] Macro("SIP/2111-b6a400b0", "hangupcall|") in new stack [Jul

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 list! On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and I'd like to have a menu point for the voicemail, but just if the user has a Mailbox. I found the AMI-Command MailboxStatus, but it does not return what I need, since it returns 0 if the user has a Mailbox but no messages and if the user has no

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,

Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you

FYI to anyone else who may be experiencing random hangups; I removed the busydetect=yes lines from the conf files on my asterisk servers, and haven't had a hangup since. I had done that once before and it didn't seem to have much of an effect, so I'm not breaking out the champagne yet. But so far over dozens of calls both made and received since I took that line out, I

Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress

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

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Hello all, HI i am very new to asterisk and my boss needs me to investigate setting up asterisk for a new client. I have downloaded and installed (make, make install and make progdocs)asterisk on my personal computer and when i try to run it (./asterisk -vvvc) i get the following output below: NOTE: i am running REDHAT 9.0 on a 796MHz cpu machine: I am excited to be able to work with asterisk

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling