similar to: Zaptel problem with pots lines

hi guys, i got a TDM400P FXS card an everything is fine except for this when i do modprobe wcfxs , the linux shows 2 TigerJet Network Inc Model 300 128k. i don't know why it is showing 2 of them. or is that what it is? ERROR: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 0 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 0 cshould be

I apparently have a dead FXS module. Is there any kind of test I can run on it (on a live system) to determine if its good or bad? The telephone has no dialtone, gets no calls. It has been working for several months, and just quit yesterday. Thanks! Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO

Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hi guys, Can someone tell me if there is a way to increase the volume of a conversation that occurs between two SIP channels or between a SIP and an IAX channel ? My headsets are set to the maximum volume but the voice is still low ... I know there is a configuration in zapata.conf for the digium cards, but is there a place I can set this up for RTP conversations ? Thanks,

The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One

I just installed an FXS module onto a 4 channel tdm thats about 5 years old and it wont work. Running dmesg I can see the following error: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test !!!!!!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!!!!!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!!!!!! COMMON_MIN_TRES iREG 1E = 0 should

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

Hi, I am interesting how can I use the capabilities of ProSLIC to measure the following PSTN parameters: - Voltage (V) & Polarity (+-); - Current (A); - Frequency (Hz). Are there any ready for use tools? If there aren't ready for use tools how can I do the above measuring? Best Regards, Miroslav Nachev

Hi All; In the zaptel, we were increasing the gain of the voice volume at the hardware level from the /etc/zaptel and /etc/modprob.conf files, but now we are using DAHDI, so where to do the same thing? I am looking actually to increase the volume at hardware level and not software to avoid the DTMF detection problem and to have better voice quality. Any advise? Regards Bilal

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] =>

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

Hi all, Yesterday, in an attempt to take back my phone room, I pulled everything apart as far back as the demarc and rebuilt it. In the process of putting things back together I accidentally connected my incoming lines to my FXS ports and my phones to my FXO ports. I quickly realized the mistake I made and corrected things but not before one of my FXS modules was smoked by incoming ring voltage.

I'm afraid that I am at a loss here. I am new to Asterisk, and have successfully set up SIP. But I cannot get my FXS card working, and I'm not sure what else I can try. # modprobe wcfxo /lib/modules/2.4.21-37.0.1.ELsmp/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

Yesterday my * server (SuSE 9.2 pro on Athlon) just stopped, no screen, no reaction to keyboard or mouse. I get all kind of messages, or just stop during restart 1. just two lines of a code (immediately after turn on the computer) 113-AM21200-100-GB GV-RX30128D F1 2. Main Processor: AMD Athlon (tm) 64 Processor 3200+ <CPUID: 0FF0 Patch ID: 0041> and BIOS line

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Hi all, Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [root@server1 zaptel-1.0.7]# modprobe wctdm ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed

Hello! I am going to use the JITTERBUFFER function in a SIP (and local channels) only setup, but have some questions of how to use it: 1. Do I need to activate jbenable in sip.conf? Or is it enough to call the JITTERBUFFER function? 2. What is the preferred way to invoke this function? Say I have channel A which is not in need of buffering, while channel B do need it. If A

> > 1. Do I need to activate jbenable in sip.conf? Or is it enough to call > > the JITTERBUFFER function? > > You only need to use the JITTERBUFFER function. > > The jbenable option will enable a jitter buffer on every channel > created for that peer (or, if global, for every peer in the system). > Depending on the version of Asterisk, it will also place the

hi, i have problem with pass-through faxing with this scenario hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.2.X(xen virtual) - linksys ATA i can fax to fax2mail on hylafax but after upgrade asterisk2 to 1.4 faxing is not working hylafax(iaxmodem) - iax - asterisk1 1.2.22 - sip - asterisk2 1.4.X(xen virtual) - linksys ATA configuration is same do you hava any idea what is