similar to: extensions => s,1,Dial(Zap/2/number) noise

Hi, in response to a previous posting regarding getting the x100p to work, I was told to run "zap show channels," but when i do i get "no such command 'zap'" There was a previous posting on this, but the guy never posted the solution. thanks, Imran

Hi, I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxs (for the fxo cards), which worked fine. ztcfg is showing two channels configured, but when I start asterisk and do show channels, i see no active channels. zapata.conf has: signalling = fxs_ks

Hi, I've purchased two x100p clones, and when I try accessing a line from asterisk with something like this: exten => _1NXXNXXXXXX,1,Dial(Zap/2/{$EXTEN}) (is that only supposed to put you on channel 2 or actually dial the # for you?) but I first hear noise, then a dial tone, but as soon as I start dialing numbers I get feedback and noise, and the call doesn't go through. Any

Hi, I've been trying to get my zaptel x100p cards working for the past week now. this is what I've done: installed asterisk: make clean make linux 26 (for fedora core 2) make install installed zaptel: make clean make make install did a modprobe zaptel, and wcfxo got this in /var/log/messages: PCI: found IRQ 11 for device 0000:00:0f.0 wcfxo: daa mode is 'FCC' found a

Hi, Sorry, in my last mail I wrote "wcfxs" instead of what I actually used, "wcfxo." I just got two digium x100p clones and installed asterisk on fedora core 2 which took some tweaking. After getting asterisk up I installed the zaptel stuff - then modprobed zaptel, wcfxo, which worked fine. ztcfg is showing two channels configured, but when I start asterisk and

Hi, I'm trying to dial out on a vonage line connected to a zap channel using stuff like: exten => 200,1,Dial(Zap/2/${EXTEN}) but it doesn't work - when i dial in the extension, i can see on a phone connected to the same line that it's gone active - but there's no dialtone. also tried adding a wait period before accessing the line and exten =>

Hi, I was wondering if anyone knows where i can find out the standards used by telco's in different countries... and how to configure asterisk to support them. Secondly, whenever i try Dialing a zap channel, all i get is no sound on the phone source, and noise on the destination line. thanks Imran

Hi, I'm trying to dial in from one phone and give it access to another line (ie incoming on zap/1 and outgoing on zap/2)... how can I transfer the call from channel 1 and give it the dial tone on channel 2? I can use dial but that takes a phone number, which I want the user to be able to select. thanks Imran

Hi, 2 days ago I've downloaded and compiled the * from cvs, along with zaptel and libpri. Now I have very strange behaviour sometimes. I call the asterisk box from outside and the zap x100p shows that line is busy and drop me right into the voicemail. But! After that It doesn't hang the line anymore. It is kept offhook. And on the top, my voicemail records VERY strange thing. first for

Hi I have the X100P card which as to sockets (LINE - for fxo line ) and PHONE (to connect a n analog line) This card is setup as fxs_ks I was getting dial tone but suddenly no Dial Tone....Help appreciated.... When I try to route the Call using - Dial Zap/1) to this FXO Line I get this error: ------------------------------------- linux*CLI> -- Executing

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

Greetings, I have Asterisk@Home 1.1 with an X100P, which is my only outgoing line. Call waiting will not work when an extension is dialed out on that line. The only way it will work is if an extension has dialed another, and someone calls in on the ZAP line. Then, the flash key will switch between the other extension and the ZAP line. I cannot get my SAP-2002 to tell the ZAP to flash

Two days ago, I was talking on the phone from the FXO, to a SIP phone. After some time (like 1h30m), all of a sudden, there's a huge noise, like a buzz... Really loud. So I hungup, and called my asterisk box again... All I could hear was that sound. Someone called me from the internet, and as Asterisk dialed the FXO, all she heard was that noise too. So, I logged in my Asterisk server,

I recently purchased a Asterisk Developer's Kit (TDM) and now have it outfitted with 2 FXO modules and 2 FXS modules. I'm not using the X100P modem card that came with the kit. I'm having problems with dialing out on my POTS line. Successful dial out is intermittent. About 50% of the time the call goes through. The other 50% it is dialing the wrong number. ( I can hear the error

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Matteo Brancaleoni > Sent: Monday, July 26, 2004 5:22 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Can't dial SIP<->EuroISDN (HFC-S based > PCIISDN card): Unable to create channel of type 'Zap'