similar to: New Installed X100p

Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: ---------------------------- /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field ?owner? specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: ***

Hi. Please help. When trying to compile Zaptel 1.4.5.1 I get the following: /build/include/linux/modversions.h -DSTANDALONE_ZAPATA -I.. -o base.o -c base.c base.c:48:29: linux/workqueue.h: No such file or directory base.c:292: warning: `vpm150m_firmware' defined but not used make[2]: *** [base.o] Error 1 make[2]: Leaving directory `/usr/src/zaptel-1.4.5.1/wctdm24xxp' make[1]: ***

Greetings to all list members! My name is Aldo Sudak and I am an Asterisk newbie. I am writing now because I have not been able to find any mention to the issues described below, neither in this list nor in the wiki. I am performing preliminary tests with Asterisk 1.4.11 and Zaptel 1.4.5.1 on Fedora Core 6. Installation of all of the Asterisk packages was straightforward and I always obtained

Hello 1. I don't have deep knowledge of either Linux or Asterisk, but I seem to have successfully installed 1.4 with Zaptel (for support for an OpenVox PCI FXO card) on a stock Ubuntu 7.04 Server Edition: ======== dmesg ========== [ 25.990943] Zapata Telephony Interface Registered on major 196 [ 25.990948] Zaptel Version: 1.4.5.1 [ 25.990950] Zaptel Echo Canceller: MG2 [ 27.523605]

hi there i am facing problem in installing the ztdummy module in fedora7, 2.6.22.5-76.fc7 is the version of the kernel. here are some logs for your kind consideration, i have tried varios solution from voip-info.org and many more, but in vain. Message from "/var/log/messages" Sep 13 14:27:14 localhost kernel: Zapata Telephony Interface Registered on major Sep 13 14:27:14 localhost

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous

The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous

Hi all, After reading great things about the OSLEC Echo Canceller (http://www.rowetel.com/ucasterisk/oslec) and seeing the reactions of people who have tried it on a recent Trixbox thread (http://www.trixbox.org/forums/trixbox-forums/open-discussion/need-people-echo-problems), it sounds like it is the "bees knees" for sorting out echo problems with cards like the x100p. Has anyone

Hello all, I am planning to setup a MeetMe conference functionality on Asterisk-1.4.13without having a Zaptel card. All users will be calling through SIP only. AFAIK, the said application needs a timer which makes use of the ztdummy module. I have basically two (2) problems I am encountering here that [1] I can't load the ztdummy.ko module and [2] Asterisk don't run when running it

Does anyone know if the Dell Power Edge 1900 has an issue with multiport E1 cards? We've had this server running for a while now with 2 E1 cards. At first we tried to install an Openvox D210P card with two E1 ports but after a couple of kernel panics we thought that maybe the card was defective and we replaced it with two Digium TE120P cards. Now the customer needs a third E1 port and

Hello All, we have Asterisk 1.4.9, Unicall 1.4.9-0.1, and Zaptel 1.4.5.1. If we were to update or recompile Asterisk, would we need to do anything with Unicall or Zaptel? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/b80fa496/attachment.htm

I installed DAHDI (2.1.0.3) on a machine with asterisk 1.4.22 and libpri 1.4.7 and I am getting the error: -- Requested transfer capability: 0x00 - SPEECH -- Called 23/317506XXXX -- Channel 0/23, span 1 got hangup, cause 99 -- Hungup 'DAHDI/23-1' on DIALING out. Calling in seems to work just fine. Seems like everything is configured fine... system.conf

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

Hi, >However, I believe that zaptel >= 1.4.6 or zaptel 1.2 >= 1.2.21 should >support hires timers for timing on kernel >= 2.6.22 . > >What version of Zaptel do you use? > I was using version 1.4.5.1 I just downloaded and installed version 1.4.7, configure/make/make install finished without an error, but when is used modprobe ztdummy the system said: FATAL: Error

Hello List! We're having trouble making call deflection on ISDN PRI. We would like to transfer a call to an external extension but keeping the callerid of the caller so it can be presented to the receiver of the transferred call. At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware TE420B. We've ordered the service (CD) from the phone company. The

Hello I've been googling for this message, but can't find why Asterisk sends a warning. The configuration files look similar to http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample It's a TDM card with just one FXO module on it, and I connected an RJ11 cable to it from the wall plug: # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv

Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 E&M wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using ZapBarge to listen to the channels themselves you can hear the dtmf digits being dialed,

Hi Friends! I need help! I'm still Asterisk rookie, so please forgive me. My Asterisk is no longer answering incoming call on the phone line. I call the phone and it rings but asterisk is not picking it up. The phone line is attached to port 4 (FXO) on my digium TDM411P card. I am running Asterisk 1.4.11 with zaptel-1.4.5.1 and libpri-1.4.1 on Fedora Core 5, Linux Kernel

Hi Sorry to use this public place, but IRC and emails to webmaster at digium have not helped in the past. I have several issues with using the files server downloads.digium.com, which has replaced the simple ftp/http file server ftp.digium.com. In downloads.d.c the directory listing is served through a seperate per-directory script with an obscure name. Let's look at

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on