similar to: Live CD - do you think they are worth doing?

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

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:

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

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

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.)

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

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

I must be getting something wrong about this call pickup. In zapata.conf I have just the default callgroup=1 and pickupgroup=1. If I call from my mobile to * and then try to dial *8 from any other phone than the one which is ringing I just get a "Nothing to pick up" answer on my * console. I also have experimented with those parameters in sip.conf but are not aware of exactly where to

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

All, For the longest time I was having this strange folder recursion problem. It persisted through many samba upgrades. The problem can be seen here in this thread: http://groups.google.com/groups?hl=en&lr=lang_en&ie=UTF-8&oe=UTF-8&safe=off&frame=right&th=48f08dabcabb2877&seekm=9oglr1%241d66%241%40FreeBSD.csie.NCTU.edu.tw#link1 I think my configuration was wrong, and

Hi, Anyone know anything about Asterisk's support for door phones? Receiving the call from the door intercom system, opening the door, etc? Any hardware recommendations? I understand that the equipment we have now is Panasonic proprietary and came with the currently deployed Panasonic TD12-32 pbx. We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while, so any info will be

According to <http://wiki.centos.org/Manuals/ReleaseNotes/CentOSLiveCD5.5> "There is a "Network Install" option on the Live CD that is the same as our CentOS-5.5-i386-netinstall ISO". I've looked quite carefully at my CentOS-5.5 Live CD (on a USB stick), and I don't see a Network Install option anywhere. Could some kind soul explain where it can be found,

Hi, I was *very* surprised by this little trick for new players: mean() only considers its first argument! > mean(1,1,2) [1] 1 > mean(2,1,1) [1] 2 I found this very different behaviour to max(): > max(1,1,2) [1] 2 > max(2,1,1) [1] 2 Perhaps this is the wrong list to ask, but does anyone else think this a little on the interesting side? Is it not possible to detect a

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

Hello. I am looking for details of the maximum allowed/usable/effective wire/cable length of the connection from a FXS port of Digium analog cards to the analog telephone handset. To clarify my intention, I need to have an analog telephone connection to my asterisk box that is 3000 meters (3km) away at least. If you have any details of ATA boxes or other similar devices that I could use to

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Reason. I have a fax/ans phone with handset, that lets you monitor the caller, so if you wish, you can pickup the call. The asterisk is undergoing testing, it will then be online tested at the house so I can get more familiar in setting components up, e.g. sip phones, voicemail, transfers etc. But, I really need the monitoring of the Voicemall being left, with the ability to pick up the