similar to: End of migration: adding support for some an alog phones

A few weeks ago, suddenly, reading news at lunch, I could not get to nytimes.com. I could ping it, and nslookup it, and if I put the IP address in place of the name, it was fine. After *much* back and forth over a ticket I put in, over the last week or so, our group figured it out: It *seemed* to be related to IPv6, and there's only *some* few sites, such as the Times, and Orbits, and one or

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's

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

Hi, I am getting half channel audio on cisco 7960? Any idea why? Details below. Jerry This same phone has been working for MONTHS using a TDM2400p with no issues. Today we got a T1 installed coming into Box A with all incoming calls going to Box B (the TDM2400P box). The TDM2400 is no longer in use. All incoming and outgoing calls are done with the T1 and Box A and Box B use SIP between them

Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

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:

Hi, I cannot make my TDM2400P (with Echo Canceller module) detect faxes. I tried with a TDM400P and it worked at 80% (20% of faxes were lost). My test conf.files are: _zapata.conf_: context = inbound faxdetect = incoming language = it musiconhold = default signalling = fxs_ks callerid = "call_or_fax" channel => 1 extensions.conf: [inbound] exten => s,1,Answer exten =>

Do the zaptel drivers need to be told >NOT< to use the software echo canceller when using a TDM2400p card with hardware echo canceller, or does the driver figure this out by itself? Thanks, Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/7dc52295/attachment.htm