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Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm

First of all great job on diax. I downloaded it and tried it, could not connect, got an authentication rejected,but I have not had a chance to figure out why yet - tried with a working gnophone setup in the configuration files. Is there any way to pass command line arguements to the program ? Where I see a real niche for a lightweight softphone is being able to serve the thing from a

Hi, Curious, do many of you check out software or projects when they have a live CD or does that make any difference to you? Does anyone know if the general public (not reading this kind of list) is attracted to a Live CD more than an Install one? thx, /r

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We have a client with a Nortel PBX with digital phone sets. Due to T1 problems (old firmware), we are interested in trying a FXO channel bank. Is there a channel bank (or equivalent) which emulates Meridian digital phone sets? In order words, an FXO channel bank that's Meridian digital? Thanks MD -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a

Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

This may be of interest -- as a tool we can use to test our systems and as a weapon that may be used against us :) http://warvox.org/ A brief read-over looks like it uses iaxclient and ruby to war dial a range of numbers and record audio samples to be analyzed to identify if the call was answered by a modem, fax machine, human, etc. The calls are placed through a PSTN termination

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hi all, There is something special I must configure in order to get the voice mssage by mail? In voicemail.conf I have: serveremail=asterisk@mydomain.ro attach=yes [default] 301 => 6535,Home Mailbox,dtoma@fx.ro I have tried to let a message for 301, but this message is not forwarded by mail. I am missing something? Thanks, Dan

Hi, I'm no phone guy but I've been having quite a bit of fun messing with Asterisk. I just unwrapped my new Wildcard TDM400P (TDM40B) which is the new 4 port FXS PCI card. It has 4 RJ45 connectors and I'd like to hookup 4 regular phones which are RJ11. If I crip RJ45 connectors to the phone cords do I simple use the center two pins (pins 4 and 5)? Thanks, Randy Smith Tiger

Has anyone successfully run asterisk with a VIA processor ? I have tried unsucessfully, do I have to run make with any specific switches ?

Hello! Several of my customers would like to add a backup to their Internet connection. ISDN is a good solution: decently fast for a dial-up-type connection, yet still faily affordable. While I was at it, I decided to look at a couple of more creative telephone service options to possibly improve their service or lower costs at the same time. These customers range from having just a

Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm

android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says "Don't Change the Format Unless You REALLY Know What

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

On 15-06-15 08:48 PM, Matt Darnell wrote: > In the past we have used Adtran Atlas 550's to break out FXS ports for > devices like modems. The great thing about the 550 is that internally > it is all TDM so there is absolutely zero latency. > > We are able to use ATA's for faxes and analog phones but devices that > use modems, they fail 99.99% of the time when using an

When immediate is set on a port that is an fxo, what is the meaning of this ? Will it go immediately to the "s" extension of the context when the line first rings, or something else ? I am looking for a way to stop the line ringing two extra times before being answered by the channel bank. (caller id is set to off I believe, and the lines in question don't have it anyway.)

I have read over the archives and am still a little confused about mgcp support in asterisk. I realize the mgcp channel is server side only (but jump in and correct me if I got this wrong.) I have seen a few references to sip<>mgcp interteroperability in patches, so now I am wondering what the case is with support in the code. What I actually want to do is have asterisk act as the client