search for: drunkcoder

Displaying 7 results from an estimated 7 matches for "drunkcoder".

2007 Sep 14
6
DECT SIP phones
Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some
2003 Apr 16
2
Globals ??
I'm interested to know whether the globals in extensions.conf is extensible ? has any thought been given to Time-of-Day etc varibility ? eg: week day after 6 change from work to home (auto transfer) ? Gary .
2003 Sep 11
1
newbie - sip, pxb, ata, nat
hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for
2003 Oct 08
1
Asterisk role
Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it
2007 Oct 26
2
Initial review of American Telecom X10001P DECT/SIP phone
Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset's speaker. DTMF is rfc2833, so what I'm hearing through the handset isn't affecting
2004 Jul 01
9
Config Files
Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot
2009 Feb 01
5
Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine
Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the