similar to: Additional Wildcard TDM2400P Setup

Hello, We use MS Access 2000 (I know, we're migrating away from it) as an application to automatically dial phone numbers. The old phone system we have allowed the call representative using the application to take their phone off hook, push a button in the app, and the app would send the phone number to the phone system and dial the number. We are moving to Asterisk for our main phone

Hello, I have a strange problem with one of my Zap channels. A user told me that he was in a voicemail system during a call, hit the Flash button, and the call hung up. Now we get no dialtone on the phone hooked up to the channel. Here's the status of the channel: jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7" Parsing /etc/asterisk/extconfig.conf Channel: 7 File

Hello, I've found a little documentation on voip-info and on the asterisk- users list, although I was hoping for an example of a tried-and-true failover setup between PRI and SIP. We are an outgoing call center that uses asterisk 1.4 connected to 2 PRIs from the local telephone company in one group (g1) and a SIP trunk from bandwidth.com. The PRIs are the primary outgoing service,

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

Hello! I am migrating an old Red Hat Samba 3.0.9 server to a new Debian Etch Samba 3.0.23d with an OpenLDAP backend. I've got almost everything working with the new server except the roaming profiles. When a user logs off, Windows complains that the permissions are not correct and the profile can't be saved. I would LOVE to get rid of roaming profiles but that isn't an option

Hello, The company I work for recently purchased 2 Rhino CB24s and a Rhino PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2 PRIs from our telco. The CB24s are for all internal analog phones. Most of the phones are setup in "batphone mode", which is "immediate=on" in the DAHDI config. They are set up this way because we are an outgoing call

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

Hello, We have finally upgraded to Asterisk 1.4, however we've run into two issues that weren't occurring before the upgrade. Issue #1: We're an outgoing call center and need to record all calls. We use the uniqueid field in the CDR to match with the recording, which we labeled with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct in the CDR. Here is the

Hello, I've been looking at some SIP packet dumps captured with tcpdump on the PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something strange, at least to me. All of the SIP packets going out from our Asterisk PBX to either of our 2 VoIP providers are consistently 50% out of order. In addition, if I use Wireshark's voip call player, the outgoing side of the call

Hello, I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out when a user gets a voicemail don't have the timezone set in the header, so they're appearing in the user's email clients at the wrong time. Has anyone else seen this? I didn't find any bug reports or other info with Google. -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road,

Hello, I may have found a serious issue with 1.6.1.6. I just compiled it yesterday on our server. When anyone tries to access the name directory through the Directory app, the asterisk process completely dies. Our extensions are in a realtime MySQL table, and the directory has worked fine with previous versions of asterisk. Jason Martin Metrix Matrix, Inc. 785 Elmgrove Rd, Bldg 1

Hi, I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my intention is to use a TDM2400P echo cancel module). It TDM2400p working good with asterisk 1.2.1? Or I need to install a new asterisk version? TIA Giorgio Incantalupo

Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 im having an error message when in running asterisk with the tdm card in. here's the error from the console of asterisk: [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get span status: Inappropriate ioctl for device [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:10466 build_channels: Unable to register channel

A big G'day to everybody on the Asterisk list. I am having a lot of trouble getting the TDM2400P card working in asterisk. I will give the important details below, please let me know if I am doing anything obvious or ideas for debugging this. SUMMARY: I get the right voltages on the line with the phone on or off the hook, but no dial tone, no ringing in or ringing out. INSTALLATION AND

Hello, Thank you for your support of Digium and Asterisk. Yesterday, we identified a problem with the driver for the TDM2400P card when used with FXO port modules. The X400M was producing (and expecting) the audio signal at an abnormally high level, which negatively impacted the performance of the echo cancelers, both hardware and software. It also resulted in unusual audio artifacts when the

Hi All, I have the problem with TDM2414E card (16FXO & 4 FXS with echo cancellation). I have already connected to the asterisk server and loaded wctdm24xxp module. When I connect the phone to FXS port, It gets the dial tone but I cannot dial to any number (I can press the phone button but nothing happens even the busy tone). There is no activity in the asterisk CLI. It just prints

For example, how much power would a server with 3 of the TDM2400P fully loaded cards draw? We're trying to figure out what APC power we should have to achieve approx 30 minutes back-up time. Would appreciate any learnings or methods on calculating the power draw and determining the appropriate back-up device. Thank you. Melisa.

I am having issues with a TDM2400P. It appears when the ZAP channel dials out, it randomly chops the first digit off of the number. I have tried relaxdtmf=yes, turning up and down the txgain, turned off and on the echo cancellation, generated new zaptel (with updated spinlock.h)... I am at a loss. Can someone please offer some help? Thanks. TJ

Hi list, I have this problem: when someone is making a call, with asterisk and a TDM2400P connected to 8 fxo lines, the sound is good, but if three, for people are calling at the same time the sound got worse and worse. Using other voip cards the sound is much better even with all user calling at the same time. What can be, the problem? Someone else has having the same

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files: