similar to: ZAP Channel doesn't hang up - X100P

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Hello All, Sorry to rehash a question I am sure has shown several time but I cannot google up the answer from the lists. Does anyone know where I can get some royalty free, cost free music for my music on hold? I saw someone's post several weeks ago that said that this exists at a download site but I have not been able to find it. Thanks! Wiley Siler -------------- next

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

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

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

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

Hi, I recently purchased from a friend 2 Cisco 7905G for testing them with Asterisk. I was able to upgrade one of them with the SIP image, the other hang up during the upgrade process and now it won't boot again. When powered up, the red and green lights keep on and the screen is blank. Does any one know a procedure to fix this ? I do not have a contract with Cisco, I have even call a

TDMoE was it. Thank you!!!! Wiley ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Robert Goodyear Sent: Friday, June 24, 2005 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Exposing Zap Channels on Server A to be UsedByServer B

You should see something like this.... (I have 8 channels) tuxpbx*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo incoming en 1 incoming en 2 incoming en 3 incoming en 4 incoming en 5 incoming en 6

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

Robert, Essentually I want to be able to have Server B dial the extensions connected to server A as well as route calls to the outbound route on Server A. Server B will have little to no knowledge of what is on Server A. I just want it to dump the calls off. For some reason I keep thinking this was a PRI type of thing. Like there was a module that loaded up as a fake PRI that your

Does anyone know if SIP will/is support on handheld PCs such as the iPaq or Axiom? With their integrated 802.11b and Bluetooth it seems like a solution to provide a wireless based sip phone for any user would be possible. Handoff between access points might be problematic but most users I know would be using their PDA phone in an airport with free wireless or at the local cafe, etc, etc... Can

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

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]#

Can anyone recommend software that will allow me to utilize my VoIP provider and send fax over IP? I use Asterisk now for my phone system. Thanks! Wiley E. Siler Director of Information Technology Education 2020 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:wsiler@education2020.com <mailto:wsiler@education2020.com>

Hello * users I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection from the Asterisk server to the outside world or any other

Hello * users Sorry I forgot to send the mail in plain text the first time... I've have a rather disturbing problem, which I don't know how to debug or how to solve, but first a brief description of the set up. One Asterisk server with a TE410P card installed (first line used on this only), and a number of Wellgate 3504A (4 port FXS devices with SIP firmware). There is no connection

Hello All, I remember there is a way to use two Asterisk servers and allow one to see a virtual trunk that makes it so server B can use the ZAP channels on server A. Does anyone know where I can find this? I am racking my brain trying to remember the terminology. It was like creating a 24 channel virtual T1 connection from server B to Server A that allowed server B to not have any ZAP

Hello All, Has anyone successfully gotten the Click to Dial to work in SugarCRM in the latest AAH? I keep getting 'Invalid Channel' but I cannot figure out why. Thanks! Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050607/b18b3743/attachment.htm

Anyone have an opinion about these guys and their recent performance? I need some local DIDs and they provide for my area code.... Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050511/365bc7b0/attachment.htm