similar to: problems with X101P and s400P cards

I am always getting multiple PCI Master Abort messages when I try to plug in a second TDM400P. I have asked this question before, but I nothing really solved my problem and I just put it on the back burner for a while. I am at a point where this is a crucial issue. I have read that the Zaptel devices share an IRQ, is this causing the problem? Is there a way that I can manually change the IRQs of

Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and

Hi! I have this configuration: SIP client A <-> NAT box A (real external IP) <-> Asterisk server (real IP) <-> (real external IP) NAT box B <-> SIP client B The echo test form any of the clients to the asterisk server is working just fine, even without canreinvite=no. When I try to call from SIP client A to B, wihtout the canreinvite=no in the sip.conf, the call

i am using RH8.0,when i try to load the zaptel driver using "modprobe zaptel" it give an error "cant locate module zaptel" also when i do "modprobe wcfxo" it gives "can't locate module wcfxo" when i try compilling zaptel i get "make ***[zaptel.0] Error 1 what am i doing wrong someone please help! -- Augustine GDES 18b oshin road,kongi

hello, I have been using asterisk now for about 2 years now on a RH8.0 it is our main call gateway. I have on the box 3 T1 TDM cards connected to 2 Rhino channel banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA 186s. It has been working good till today some few hours ago. i just discovered that there were no dialtone on the phones. Asterisk did not spit out any error, it

hello * users, i am using a zplex 10b(8fxosand 16fxs) connected to a T100 digium card, RH8.0. soon after running * i get this warning "file chan_zap.c,line 4155 (ss_thread):CallewrID returned with error on channel 'zap/7-1' then executes the other applications itself. --executing Wait("Zap/7-1","1") in new stack. -- executing..... it gives the same warning

I am about to try our TDM400P E model from the developer kit (not the "Lite") we just got and noticed a large number of reported problems. I had the CVS from Sep 12 (or so the CVS/Entries file has in it). My drivers seem to modprobe fine. My card show up as "Found a Wildcard FXS: Wildcard S400P Prototype (4 modules)" in dmesg and no apparent errors...yeah. Is "S400P

HI, I just installed an additional module to a TDM400P card. Everything worked fine for a while, but after rebooting the server the new channel fails to be recognized by asterisk. Here is the output == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling -- Registered channel 2, FXO Kewlstart signalling -- Registered channel 3, FXO

Hello Dovecot users, Here's the Pigeonhole release that goes with Dovecot v2.3.1. You will need this release for Dovecot v2.3.1, because the previous v0.5.0.1 release will not work. Apart from compatibility changes, it only contains bugfixes. Changelog v0.5.1: - Explicitly disallow UTF-8 in localpart in addresses parsed from Sieve script. - editheader extension: Corrected the stream

Hello Dovecot users, Here's the Pigeonhole release that goes with Dovecot v2.3.1. You will need this release for Dovecot v2.3.1, because the previous v0.5.0.1 release will not work. Apart from compatibility changes, it only contains bugfixes. Changelog v0.5.1: - Explicitly disallow UTF-8 in localpart in addresses parsed from Sieve script. - editheader extension: Corrected the stream

Hello everyone! I installed Asterisk following the instructions of the book "Asterisk: The Future of Telephony". (very nice book) However, I failed. I installed zaptel, libpri and asterisk (in this order). The Installation of Zaptel is successful and my TDM400P is correctly detected: # zttool Alarms Span OK Wildcard S400P

Let me start by saying no, this is not the normal stupid echo question :) I currently have a SIP device and an X101P, and have had the usual echo issues and have played around with the various solutions, none of which are quite perfect (understandably).. however, my girlfriend is a little more picky than I am when she uses the phone, so I'm considering a Digium TDMXXB card for the house. My

All, I'm having some issues with an X101 clone. The machine is not plugged into the network right now, but I'll pull the configs and send them on shortly, but they are similar to the sample configs. Problem 1: "ZAP cannot create channel" For some reason, the cards hang after a call. It's pretty annoying since the only thing that will fix it is a restart of asterisk.

I'm having many troubles with x101p (orginal from Digium, wcfxo kernel module) on analog line simulated by a Cisco Router I'm experiency random hung up while zaptel doesn't recognize call progress (Italy signalling) Signalling is simple ignored as if someone hangs up on the other end of the analog line, on the asterisk end is it possible to listen the busy tone as generated from

Hi! I've bought a Wildcard X101P card to play a bit with asterisk. I'm just trying to place a call to a number, wait until the receiver takes the call, and then play one of the .gsm sound files. I'm in France, so first I tried modprobing wcfxo with opermode=1, but the DAA mode remained FCC, so I changed the opermode variable in wcfxo.c to 1 by hand and recompiled & reinstalled

Any other FreeBSD users experiencing this particular problem? I have an X101P board plugged into a PC running 5.2.1 and using the latest (0.6) zaptel port right from the port tree. zttool shows a RED alarm on the card but the phone line appears to be okay (plugging an analog phone into the "Phone" port produces a dialtone on the phone). I've even tried switching the line polarity and

Hi, I'm new to *. I just installed my X101P card with * from the source on Mandrake 10.0 and I test it. Everything seems to work fine. When I call at my home office all the demo ivr seem to work. But I have one question regarding * using /dev/dsp0. I only have one sound card on my system and it has to be use by my personnal PVR called MythTV. I though that * did not need a sound card to work.