similar to: ZAP channel (X100P) won't detect call waiting

Hi, I have already installed the X100P drivers but when I make to make call call, I always get No channel type registered for 'Zap'. Any idea how to solve this problem? It seems that the Zap module is not loaded but I have already load the driver by modprobe zaptel > modprobe wcfxo > ztcfg -vv thanks -------------- next part -------------- An HTML

When receiving an incoming call, I get sent to my IVR just fine. My Playback event plays back my test file and the itis suppose to Hangup. The Hangup app fires off and I see the console say it hungup the line. However, I cannot receive anymore calls after that. When I run 'zap show channel 1' I see my zap channel in a state of off hook. The only thing that fixes it is stop/restart

Hi, I have the following basic configuration: - one X100P card - two IP Phones (connected to an ATA186) When an incomming call on X100P, both phones connected to an ATA186 rings (configured like that in extensions.conf). When a call is established between the two ATA phones and a call occurs from the PSTN line, the callerid is correctly displayed on both phones (as internal callwaiting

I am running CVS-09/11/03-14:03 on Redhat 9.0 Trying to get call waiting / call waiting callerid working. The setup is: X100P asterisk -> ATA 186 MGCP --> analog phone. What changes to I need to make to my mgcp.conf and extensions.conf file to allow answering of Call waiting calls? And how do you answer call waiting calls with the system. I have usecallerid=yes, hidecallerid=no,

Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I

Asterisk doesnt answer the ringing phone, so according to some doc that I read I should try and see if I get I/O on /dev/zap/1. I dont see any from cat /dev/zap/1 while it is ringing or otherwise. Dmesg seems to have a few odd things (out of space), not sure how relevant. Does anyone have any ideas? Here is my info: <built zaptel / asterisk CVS from 2 days ago> [root@homestar etc]#

If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to

Hi! The X100P is working - partly. I can make outgoing calls. But the card has got a problem detecting incoming calls. Even in verbose mode I don't see any hint that the card detects a call. At the moment I'm using the card behind my old ISDN telephone system. Could it be a problem with the line voltage? Can I adjust anything when loading the kernel module? Bye! Michael

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

I have a Vonage acct that uses the Cisco ATA186. Currently, I have the ATA186 plugged into a SPA3000 to act as the FXO port. I installed a X100P card with the idea of replacing the SPA3000. Now, when I plug in the ATA186 into the X100P card and make a call into the system (from cell phone) and hangup when the IVR is playing, Asterisk is not detecting a hangup and keeps looping the IVR. If

Has anyone tried to use * and an X100P in Spain. I have enabled busydetect in zapata.conf but still its not detecting busy correctly. I guess that this is because the busy detection routines are looking for different tone sequences and frequencies. Anyone any ideas ? Rgds, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know if there is a zapata.conf option to tell * to listen for a dialtone before dialing? I've got a couple of analog phones on a pstn line shared with a x100p * fx line. If someone is on the analog phone and another person initiates a call through * to use the same line, * dials over the top of the existing conversation. Is there a way to have * detect dialtone before dialing?

I finally got round to installing a second X100P tonight in my asterisk system. Hangup detection, which was already pretty dicey in that it only seemed to work when the other side sends a busy, now doesn't work at all. I'm running the CVS-03/28/03-00:03:08 version. I have tried every combination of callprogress and busydetect and nothing makes a difference. I wonder if others in

The subject says it all... I have an X100P and I have (for now anyway) Bell Canada's Call Answer which will notify you through one of those nifty CLASS signals that you either do or do not have voicemail. This is not only a "stuttering dialtone" but some actual signal passed so CLASS-aware phones can detect it and flash their message waiting indicator. I'd like to detect

I'm running * on two pstn lines (x100p cards) that happen to also have analog phones installed on the incoming pair for backup (until testing is complete). If someone is talking on a pstn analog phone and they talk loud enough, * senses the voice (apparently assuming the line is ringing), and rings an appropriate SIP phone (7960). Is there a way to config * so as to only process inbound

Hi, I hope someone could help me, i have a x100p interface for testing purpose and on each incomming call I redirect the call to handytone 388 atas, the problem comes when i'm during a call and another call comes in, i hear the call waiting beep (comming from the zap channel), but I can't catch the call as usually using flash+2 (my pstn call wait sequence), because when i flash the

Just to make sure this isn't a typo in your original email... Is this example from your zapata.conf? Also, the extension you have shown are in extensions.conf not zapata.conf correct? Here is an example of a good zapata.conf.... [channels] language=en busydetect=yes faxdetect=both busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes useincomingcalleridonzaptransfer=yes

This may seem obvious or silly but have you tried using a different phone cord? A bad phone cord played havoc on a colleague of mine during initial config. Fro what you show as your output from ztcfg, you should have one channel configured successfully then. Your example shows you have the channel set to one so no problem there. If you run ZAP SHOW CHANNELS from the CLI what do you see? W

Hello. After doing much web searching and searching archives of this mailing list, I see that my question has been asked at least 6 separate times but no answers have been attached. In a nutshell, is there a way for a SIP phone to easily hook flash a ZAP analog trunk mid-call? (This is important when trying to make use of features on a PSTN analog line such as call waiting, call forwarding,

Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles