similar to: Digium cards just for timing

I have a normal backup phone (and an alarm panel) sharing the POTS line with the Digium X101P FXO: | | Wall |>---+------X101P FXO as Zap/5 | | | Phone & Alarm Whenever the Phone is used, Asterisk sees a 'false ring' signal immediately when the phone is hung up. The Alarm panel dials out nightly at around 1AM, and each time it completes the call, Asterisk

Sir, Sorry, I can not return your email immediately. I am out of office from May 18 to June 6, 2006. thanks for your patience, Jay Huang

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

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

Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users

This might be a newbie question but I'm just wondering how would it be possible to have 30 analog lines using asterisk for PBX by just using TDM40B and X100P (or are there any device>), if an ordinary PC support just 4 PCI slots? the maximum scale i guess would just be 2 x 8. Adding a new PC just for this purpose would be costly. I would appreciate your comments. Thanks.

Hi, I try to use Speex codec into Win32 platforms. However, I find the CPU bandwidth usage is very heavy on a Pentium 3 machine. Compare to Microsoft's G723.1 codec, speex 8k is using more than 20% cpu bandwidth. Does anyone know what is the best version of Speex to "beat" the Microsoft's G723.1's on CPU bandwidth usage? Does Speex have MMX-enabled codes? thanks very

Hello, Anyone? I have hard time to believe this is not possible with chan_pjsip. Anyway, may I ask how people handle the following scenario which I imagine should be quite common: - I have internal extensions talk to each other using g722. so their codec setting (with chan_sip now) is "allow=g722,ulaw" - I have carriers trunks that handle ulaw only (allow=ulaw) - calls between

Has anyone been able to sucessfully use 2-4 X101P clones with Asterisk on a single system ? I'm using one Encore MD 3200 modem sucessfully, alongside a TDM400P (3 FXS, 1 FXO) and would like to be able to use up to 4 X101P on a single system. In most cases I'll have 2 or 3 instead. I understand all the issues with interrupt sharing and PC motherboard quality. Just need to know if any

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

Hi, Is it possible to use 2 B channels simultaneously with either I4L or CAPI drivers? We use AVM A1 (Fritz) PCMCIA with I4L driver and AVM B1 PCMCIA with CAPI driver. Thanks, Michael.

I have a couple of questions I have looked everywhere for the answer and cannot find. 1) Musiconhold. I have turned on moh in musiconhold.conf, I have musiconhold=default in zapata.conf, and I have a Wildcard X101P thats working fine. If I connect to the musiconhold, I hear the audio, but it's very distorted, like it's over-driven. I'm running gentoo, and I emerged mpg123

I've merged some changes from Michael Manousos that should improve sound quality on non-zap channels, including music on hold. I'd like to hear back on or off list about your experiences with the new code. Thanks! Mark

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

This module wont compile can anyone give me any assistance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040614/03ae433c/attachment.htm

> 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.

dear all, i am looking to enable CALLERID on an Asterisk system comprising a X101P FXO interface connecting to BT PSTN in the uk seems this is supported by the interface but there seems to be varying information on how to enable it in zapata.conf 1. usecallerid=uk 2. ukcallerid=yes being two of the configuration statements offered TIA GT

On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actually follow the described

Then if let say instead of buying TDM400P cards I get this : Clipcomm CG-410 Quad FXO Gateway is it any good? They also sell Quad FXS Gateway. Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm wondering if it's working fine with asterisk. I found this one too but at a lot higher price : AudioCodes MP108 8-Port FXO Analog Gateway (SIP) I need to buy a

Hello all, This new version has more options (account code, AMA flags, lib tracing) in the config file and some improvements in the build process. The code is available from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael Manousos