similar to: Message from * console.

I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy ERROR[16384]: File chan_zap.c, Line 4781 (mkintf): Unable to open channel 1: Device or resource busy here = 0, tmp->channel = 0, channel = 1 ERROR[16384]: File

Hey all, ? I'm trying to get an E400P and 2 X100Ps working together in the one box and don't seem to be having much luck. ? I can get the two different types of boards working separately, but not together.? I've made calls on both the X100P and have seen sync and correct signalling on the E400P.? But when I try to enable to configs together I get the following: ? modprobe wcfxo

Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link. There is somebody he know (has experienced) how many concurrent call (Classical Phone->Voip) can handle Asterisk ? Thanks ! -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

Hi All, I've had a great time exploring * on & off over the past month and have had a deal of success, very impressed with this product... I have an issue on a debian unstable system with the zap interface. I've compiled and loaded the ztdummy successfully, but when attempting to connect to a conference room I get failure. I get the error chan_zap.c:757 zt_open: Unable to open

I just learned about Asterisk yesterday. Besides all the cool PBX stuff that asterisk can do can it also do these: 1) Receive FAX's ? 2) Control (serve) pppd dial-in connections ? 3) Is it possible to use voice modems as the FXO/FXS cards ? Thank You -- Peter http://tkvoice.netfirms.com/

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

I've put a t100p in our * server and I'm having trouble configuring it. It is directly connected to an Adtran TA 750 channel bank with two FXO cards (8 analog incoming lines total). I'm able to insmod and modprobe both zaptel and wct1xxp with no trouble, but when I start * with /usb/sbin/asterisk -c I get the following output: [root@rosella root]# /usr/sbin/asterisk -c Asterisk

I'm going to install asterisk with four digium cards. Can anyone mention a brand that carries boards with 4 compatible pci slots? Thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Dear All, I'm having a problem with my Asterisk + E100P Installation in UK (BT PRI). The system functions as expected, and my dial plan works as expected. 30 minutes (or so) after starting the asterisk service I lose the PRI line, and only get this back after a service asterisk restart or reboot. During the failure there is no alarm on zttool, ztcfg show all 31 lines and there are no

Hello, I recently started playing with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives. 1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this? 2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

Help: Checkout as of 17:00 UCT Does anyone know if: chan_zap.c: In function `zt_train_ec': chan_zap.c:1081: `ZT_ECHOTRAIN' undeclared (first use in this function) is expected at the moment? Dave Kitchen

Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'