similar to: trunk timing on 2.6.x

Hi, I got my * working fine with FWD at office with 2 extensions, i receive calls and i can make calls thru FWD. I got also my * at home, and i connected it using auth=rsa. From my home, i can make calls using my office iax, but if i try to redirect incomming calls from FWD to my * at home, it rejects the call. I created the pub/key pairs for rsa and its working ok and i just pasted the

Hi, I just updated to FC5, and used zaptel-1.2.1 as it was with my last version, it show this error: CC [M] /usr/local/src/zaptel/wcusb.o /usr/local/src/zaptel/wcusb.c:1452: error: unknown field ?owner? specified in initializer /usr/local/src/zaptel/wcusb.c:1452: warning: initialization from incompatible pointer type make[2]: *** [/usr/local/src/zaptel/wcusb.o] Error 1 make[1]: ***

Hi, I am a newbie in asterisk, i could compile it and run it with no problem on a RedHat 9. In the same box, i got a linejack and a phonejack cards and i downloaded the CVS driver from quicknet. This 2 card were working in a openh323 (openphone and pstn) project with gnugk on a RedHat 9. I am using the default samples, and i tried /dev/phone0 and /dev/phone1, but when i run asterisk, i get this

I'm looking to handle 3 PSTN lines with my Asterisk server. I have a Digium TDM22B and Sipura 3000. The Sipura works great, but the TDM22B seems to have terrible problems with my board---virtually all peripherals need to be disabled in BIOS, and then there is terrible noise, terrible silence and virtually no ability to use DTMF in IVRs. I really wish the TDM22B worked, because I much

I'm having problems with a TDM22B. The FXO modules work fine. Both FXS modules fail to initialized. The error messages I see when the module loads: Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. I already RMA'ed a TDM22B because of this problem. Now that the replacement shows a similar issue, I want to see if anyone else can think of something to

I just got my TDM22B, but the package did not include anything else, like datasheet. 1. Where can I download it? 2. Which outlets are FXS/FXO ??? (depending on the green /red card positions) 3. Does it need the extra power?? bye Ronald

Hi, I am using asterisk 1.4 with Digium TDM22B card. My system is running well except CALLERID. I have tried all options for cidsignalling, cidstart but no luck. Btw, I am living in Malaysia. From google web site, I found some one having the same problem with new version of asterisk but not in old versions. I do not want to try the old versions of asterisk. I really appreciate if someone can

Im getting the following error when I try to start Asterisk WARNING[16384]: chan_zap.c:765 zt_open: Unable to specify channel 1: No such device or address Nov 29 21:40:43 ERROR[16384]: chan_zap.c:6195 mkintf: Unable to open channel 1: No such device or address I'm using a TDM22B card. ============= # Zaptel Configuration File # loadzone=us defaultzone=us fxoks=1-2 fxsks=3-4 ==============

Is there a way to test if the motherboard is ready for a Digium card (PCI 2.2) ? I would like to know from a remote computer, where I have (root) access, if this computer is ready for a TDM22B. bye Ronald

Hi All; If one of my FXS port damaged at TDM22B because we connected the Telephone Line cable to the FXS port while it should be connected to the FXO port, then can I order S110M FXS Module and fix it instead of the damaged FXS? (This if we assume my problem that really the FXS port damaged). Rregards Bilal

Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I

I'm debugging a TxFax problem whereby the fax transmission fails. I suspect interrupt latency--some interrupt routine is holding its interrupt too long. I have all unnecessary services switched off and X is not running when I perform these tests. Some transmission are successful while others fail at random points. I've noticed that after I boot Linux, load zaptel, wcfxo, and wcfxs,

Hello. I have this problem. In my asterisk box, I was running debian woody with asterisk package from backports.org. Last friday I upgraded from debian to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686, rebuild zaptel kernel module and also upgrade to asterisk 1.0.2. But now asterisk won't start. Here is more info #asterisk -vvvvvvvv (last lines) [chan_zap.so] =>

I need to reopen this discussion because it's impossible to run spandsp (and VoIP) under these circumstances. With zaptel unloaded, I see the following "vmstat 1" output: no swapping, an occasional disk output, +/- 1003 interrupts/sec., less than 10 context switches/sec., CPU idle 100%. A very quiet system. I load modules zaptel and wcfxo, and the system utilization stays the

Hi All, I have the problem setting up TDM22B card. Steps what I have followed are: [1] compiled zaptel-1.0.9.2 & installed the same. [2] modprobe wcfxo /lib/modules/2.4.20-8/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg

Hi, I have just discovered (a year after it was implemented) a possibly undocumented incompatability between IAX in Asterisk 1.4 and any version of Asterisk pre-March 2008. It seems an AST_CONTROL_SRCUPDATE frame type was added (in March '08), but no mechanism to negotiate whether it can be sent to the remote end, so if a "new" IAX endpoint sends it, and the remote end ignores it,

Hello, I've got 2 analogue lines (from SBC) coming into a TDM22B. SBC have put rollover from the first to the second line. The rollover works fine when handsets are connected directly to the lines (ie when Asterisk is not involved), but when the lines are connected to Asterisk, the rollover fails: the caller just hears the line ringing, and the person on the first (busy) line hears call

I have a TDM22B (TDM400 PCI + 2xFXO + 2xFXS). One FXO is connected to the PSTN. When Asterisk places a call, it dials using DTMF. If I listen in on the line during the dialing, there is a roughly one second pause between the penultimate and the final digit -- proportionally much longer than the pause between the preceeding digits. This accounts for about 25% of the time taken to dial. Anyone

Hi, I have two analog phones connected on the digium tdm22b; I can't put calls on hold by pressing the R button on the phone. I can do it only by hook flash. How can I configure asterisk to use the R button? Thank you in advance.

Hi all, I have installed a TDM22B on an IBM xSeries 220 with FedoraCore 3 and setup asterisk cvs head to work properly. A few days ago, I tried to update the system kernel to 2.6.12 from 2.6.9 and also to change to asterisk stable 1.0.9. After compiling zaptel and asterisk, i loaded zaptel and tried to load the wctdm module but this failed with the following message: Notice: Configuration file