similar to: x100P: Ring/off-hook in strange state 6 on channel1

I'm proposing an asterisk configuration and considering the use of multiport SIP/FXS gateways (instead of T1 cards and channel banks). I'm looking for products similar in function to the Cisco ATA-186, but with more ports. I've seen the manufacturer's web pages for the Audiocodes MediaPack (http://www.audiocodes.com/) and the Mediatrix (http://www.mediatrix.com/) access devices.

I've tried searching the archives but come up empty. I'm looking for a "receptionist" application, that could be used in a business center. Lots of different incoming phone numbers (200+), all need to be answered in a different way, by a human. This is typically enabled with a terminal of some form that pops up the company and contact information for the called number, and a

Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine, zttest comes back with configured. If i call a line when zttest it shows on the display,and then goes when the line drops. In * when a call comes in, it follows my dialplan and answers the call according to the log, but IT DOESN'T actually pick up the call, i.e. it continues ringing. I'm using KS signalling, and

Using Digium's "Asterisk Developer's Kit (TDM)", I've been trying to make an outside call by copying sample.call to /var/spool/asterisk/outgoing. I want the VoiceMailMain to run when the call is answered. The call is made correctly but, as you probably know, the application starts as soon as the call is made. I see there are two solutions: Using callprogress=yes in

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

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

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

I've had a (small) working asterisk with X100P and 4xFXS TDM card for a while now. I've been running the release from (I think) April 13 - asterisk 0.9.0, zap* 0.9.1. I tried 1.0 RC2, and my internal FXS line doesn't ring - call comes in the FXO, gets answered, and rings through to the FXS extension, but the phone doesn't ring - asterisk shows it ringing and eventually punts to

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

>From what I've been reading about the callprogress option, it seems like it will work properly only with a T1 or PRI in the US. Is that correct or are there still issues with call progress detection even if those qualifications are met? Thanks, Shaun Tierney

I have this in extensions.conf: exten => 1,1,Dial,Zap/g9/4439568899/|24 exten => 1,2,VoiceMail(u8004) and this happens: -- Playing 'js-joe-trvm' -- Executing Dial("Zap/2-1", "Zap/g9/4439568899/|24") in new stack -- Called g9/4439568899/ -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Nobody picked up in 24000 ms -- Hungup

Hello there, I tried to found documentation about openloop disconnect on Asterisk/Zaptel. And up to now, I didn't find anything. Is openloop disconnect supported by zaptel/wcfxo drivers? Other question: maxsilence and silencethreshold don't seem to have any effect on voicemail. Could it be possible that in certain situation it just wont work? I tried adding some verbose in

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.

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

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,

I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it?s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on

Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and

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, 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