similar to: turn off dial tone on a TDM400p channel

i've already tried to change the indications.conf to the following: dial = 0/1500 but the dial tone still persists i am using the following workaround but obviously not a clean b'cos it just replace dial tone with some other tone. in zapata.conf context=spec immediate=yes signalling=fxo_ks channel=>2 ; TDM400p-1 in extensions.conf [spec] exten =>

Hi, I recently installed a gentoo on a PC, purchased a TDM400 and installed asterisk. Kernel is 2.6.11-rc3 First, there seems to be a problem with my system: I must put the pci=routeirq to make the card detected. The modules are loaded ok (dmesg output) Zapata Telephony Interface Registered on major 196 ACPI: PCI interrupt 0000:00:08.0[A] -> GSI 16 (level, low) -> IRQ 16 Freshmaker

as my first project with *, i would like to replace our old neax2400(sds) with an * server. i've got an X100p and a TDM400 on hand already. for the CO lines, the X100p works ok with fxsks signaling though there are still strange things happening every now and again but more testing is on the way. my real big problem is the DID lines which our telcos call DDI lines; (incoming calls only) i

This is, hopefully, just a case of brain fade. With zapata.conf and zaptel.conf in place, asterisk loaded, no dial plan and all LEDS on the card lit, I get no dial tone, plugging an analog phone into ports 1 or 2, only a buzz and click. zaptel.conf - defaultzone=us loadzone=us fxoks=1,2 fxsks=3,4 zapata.conf [channels] signalling=fxo_ks language=us context=phones-1 group=0

Hello, I just setup the Asterisk to integrate with Panasonic legacy PBX. Config as followings, PSTN <-- PanasonicPBX--TDM400P(FXO)--AsteriskPC --> Internet * is for AA / Voicemail and VOIP in/out Currently the AA / Voicemail function for incoming PSTN calls are working well. My problem is for the incoming VOIP call. It can ring my internal extensions and talk without problem. But

Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten => 123,1,PlayTones(ring) exten => 123,n,Wait(5) exten => 123,n,Playback(demo-congrats) exten => 123,n,Hangup() exten => 321,1,Dail(LOCAL/123 at test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice

I've setup * with TDM400P w/1 FXS, 3 FXO modules. I've one analog phone connected to TDM400P FXS module, 1 PSTN line to one of the FXO module(ZAP) , and 2 analog phones connected to Sipura 2000 (SIP). The calls between SIPs and zap phone (fxs) are OK. But 2 issues cannot be solved: 1. To dial to PSTN via zap phone, the setup in extensions.conf with the following exten =>

New system install. At what point, in bootup, should I be able to get a dial tone on the phone ports on a tdm400p? There are two fxo and two fxs ports. I know which to plug into <g>. At boot up, as soon as wctdm is loaded, all the ports "go green, yet I do not get a dial tone on the phone ports. I thought as long as zapata.conf is correct, the board should be

Is the a way to turn the ring tone OFF during dialing? When I'm in a macro mode I have to listen to ring the tone for 20sec before macro finish and I get connected. -- Joseph

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

Hello, I have an Adit 600 (3 FXS cards) hooked up to a digium T1 card in my asterisk box. I 'connected' the slots to the a:1 T1 interfaces via the command line. The slots (3 fxs) are configured with 'ls' signaling. I configured the T1 card with the same line settings as the T1 interfaces on the adit and I get green lights on both the T1 card and the T1 interface on the adit (so

Hi, I installed a 1-port FXO on my Ubuntu 8.4. I was earlier only hearing a fast clicking sound and now I am not hearing any dial-tone. The FXO card has 2 slots: [phone | line ]. I hve connected the wall-phone-input to the "line" slot and "phone" to my home-phone. I do not hear any dial-tone on my home-phone. Asterisk seems to recognize my hardware....here are the relevant

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

Hello all, I just installed a TDM400P with 2 FXO modules on my asterisk server. The card works perfectly. To get users to ring out from my SIP phones i setup an extension with 0 that basically does something like this: extension => 0,1,Dial(ZAP/g1) where g1 is the group of the two FXO channels extension => 0,2,Hangup This works exactly as i want so users basically can dial 0, wait for

Hello, I have an issue with Digium TDM 400 card series. When I try to make outgoing call (PSTN call) for example, the Zap channel could not be created and busy channel message appeared. Below is the full log : [Feb 5 09:26:17] VERBOSE[3047] logger.c: -- Executing [s at macro- dialout-trunk:20] Dial("SIP/213-09648720", "ZAP/g1/08170709XXX|300|") in new stack [Feb

Hi I've had an asterisk setup for the past 15 months, based on the debian asterisk packaging. Until late August of this year, I had no problems once initial setup was complete- the system worked essentially flawlessly. Since August I have been having exceedingly infuriating intermittent problems that are causing me occasional periods of nasty trouble: 1. No Dial Tone. Every Sunday night at

I just got the X100P and the TDM400P with one module on it, I had installed asterisk and confirgured some file, but I can't get a dial tone on my analog phone. can someone help? Regards Leo

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be