similar to: FC3, TDM11B (DEVPCI) and asterisk

Hello , I'll just started with asterisk and I would liket to to dial between your two phones with to cisco ATA 186 , but I have a problem The two cisco ATA and the server in the same networks and i have the ring in the phone but i'am not able to place a call Between the twe phone . In attachement the sip.conf and a log file Any suggestement . Regards RAbii

Hello! Over the last couple of days I?ve been trouble shooting a strange problem with Asterisk. First of all, I should say that I?ve been running Asterisk on a Fedora Core 3 box since last May, but decided to do a reinstallation of everything due to some problems we?ve had with echos during conversations (100% SIP based, so no ZAP echos). We are talking about a low-volume installation

Hi all, I have been out of the linux / unix loop for a good three years now, and a lot has changed.. I have a windows 2003 server on my home office lan as my domain controller. I would like to know if there is a good how-to that applies to Centos5 to achieve the following: Joining the centos machine to the AD domain (I believe I have done this already) Sharing files from it

Hi, we're testing Asterisk 1 RC 2 behind ordinary router and NAT. Since we're sharing network with web server it seems like voip packets are not coming through fast enough (Digium demo dies after few seconds...). It's the same if I make direct calls (passing Asterisk) so we conclude it's network problem - it also work normally outside our router... I wonder what solutions can we

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

I ported Jeff Banfield's Rweb (http://www.math.montana.edu/Rweb/) to Windows. In the process, I dropped some things (non-JavaScript version, Rweb modules and data sets), but also simplified the installation process by removing the dependency on NetPBM. A Ghostscript dependency remains, however. See http://www.liquidmarkets.com/m.php?m=News&file=article&sid=6 for up-to-date

At 1:07 PM -0700 on 4/5/04, Scott Laird wrote: >[snip] >2. Is anyone working on SRTP for Asterisk? Are there any SRTP >clients out there? > >Scott Yes to both. http://www.voxilla.com/modules.php?op=modload&name=News&file=article&sid=56 http://www.dslreports.com/forum/remark,9818470~mode=flat Read carefully. Sipura is offering crypto in their newest (yet to be

Hello all, So i am new to asterisk and very green when it comes to Linux, so don't beat on me too bad :) I just set up * on Red Hat 9.0 last night... everything "seems" to be configured coffectly, I can start * no problem and get the CLI prompt... now here is my question... I have an account set up with VoicePulse Connect! and also have a Sipura Spa-2000... i am trying to get make

Hello all, i would like to configure TDM11B with Asterisk, if any one have the configuration steps please provide me it. Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050208/1f6dfabe/attachment.htm

Hi Asterisk Community, I have a development with a TDM11B in it. I am trying to connect this to a exchange line as well as a UK telephone and are looking for some pinout information for the FXS port. Is it the centre 2 pins that at the tip and ring. I have been digging around the Digium site but cannot seem to pick up this info. Any help appreciated. Thanks -- Gary

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Dear All, All these days I was ahpppily using Asterisk with TDM11B, but from today all of a sudden asterisk has started acting strange. The telephone device connected to channel 1 rings continously, following info is displayed on console -- Starting simple switch on 'Zap/4-1' Jan 28 15:38:58 ERROR[77024176]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-8) Jan 28

Hi list! I'm not getting incoming CallerID in The Netherlands on my TDM11B. Everything was configures according to the docs at digium.com. The error on the console is this: Feb 13 16:49:40 ERROR[16123]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-84) Feb 13 16:49:40 WARNING[16123]: chan_zap.c:5396 ss_thread: CallerID feed failed: Success Feb 13 16:49:40 WARNING[16123]:

Hi all, I need help in disconnect supervision. Im running on AAH ver.2.5 at home with TDM400P with 1 FXO and 1 FXS (TDM11B). I have implemented DISA on AAH for origination (PSTN to VOIP bridging). I'm facing problems with disconnection supervision. My calls are not getting disconnected at times and it causes a lot of loss as the provider is charging me. After some serious study i have found

Hi list! I have a TDM11B (DevKit). When I start asterisk everyting is working fine. However as soon as I do a RELOAD from the console the FXS port stops working. I have a phone connected to it, when I pick up the phone after issuing a RELOAD I get this on the console: Jun 1 20:27:38 WARNING[14316]: pbx.c:2357 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension 's'

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

I have a TDM400P with 1 fxo and 1 fxs module(TDM11B) i just wanted to know how to configure the zaptel and zapata & extension conf to work well Please help me on that

I have fixed this before, but I cannot for the life of me remember how I did it. I have a TDM400P with one fxo module and one fxs module. I setup Asterisk @Home and everything works fine, except when I try and call out. If I call out with a SIP phone it calls the zap extension and not the pstn line? If I take the zap extension offhook and call with the SIP phone it dials out the pstn line

Hello all, This is my first post to this mailing list. I've been following the development (not as a developer, but as an interested user) of Asterisk for over a year now. It's only in the last month that we've decided to give it a try (hey, might as well...everything else we use is open source). We ordered a TDM11B (1 FXS, and 1 FXO). The card and Asterisk install was uneventful.