similar to: Status of FAX

Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks

Hi, im running shorewall 2.0.16 with centos 3 (iptables v1.2.8), everything is working fine for several days, i have configured a masq lan and all the outgoing traffic is ok, but now i want to redirect (port forward) the external web traffic to an internal machine, somethig like this INTERNET ---------> SHOREWALL -------------------> INTERNAL_MACHINE [public

Hi People, what files is processed first?, balcklist or rules, i want to globally filter imesh, but at the same time allow managers to connect, i.e. , imesh work on port 1214, i have this: /etc/shorewall/blacklist #ADDRESS/SUBNET PROTOCOL PORT 192.168.0.0/16 tcp 1214 192.168.0.0/16 udp 1214

Hi list, I am a happy user of shorewall, i have followed the instructions in the shorewall''s web site relative to the squid transparent proxy configuration,all works ok, but i have been instructed to let adicional specified ports (aplications) to be snated (allowed to run) together to the web browsing service,i mean if i snat the network (i have a static ip from my isp), all

Hi people, is there any good document on setup asterisk using only its voip functions , i mean without e1/t1 hardware, i have a cisco 5400 gateway and a cisco 7000 working as gatekeeper, can asterisk register with this GK?, im thinking in using cisco voip phones Any pointer will be apreciated, and sorry for my bad english Miguel

Hi I have a trouble with centOS 4 The process kswapd0 eat almost all my memory space, and also this process get almost 100 percent of cpu, and the box go slow I have read that this problem is fix when the kernel is upgrade but I do not know hot to update my kernel. when I type yum list \kernel* I get the following list: [xxx at xxxxxx]# yum list \kernel* Setting up Repos addons

Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to

Hello, I'm not a very experienced Samba user. Now I have the following problem: I have implemented a document management system (with version control tool ClearCase) running under Unix. All the NT-users are also available under unix with the same name. The directories containing the documents are exported via Samba. For accessing the documents from a WEB-browser running on NT and for

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

Can anybody share information how to install NVBackgroundDetect? I have the app_nv_backgrounddetect.c but I'm missing: app_nv_backgrounddetect.o and app_nv_backgrounddetect.so and have no idea how to generate them. Wiki point to contact Newman Telecom but all I received was the app_nv_backgrounddetect.c and no instruction how how to install it. The installation instruction from wiki

Hi all I have a box with Centos 4 and qmail with all pathches. since Last month i have had twice the same problem. The process kswapd0 uses all my resources of CPU and RAM memory and the consequence is that my local queue grow up and the box go slow too. The first time i guess the problem was the version of the kernel, so i did an upgrade form kernel.i686 2.6.9-5.0.3.EL to kernel.i686

Hi Everyone. Some time ago, I was told that it's possible to implement an incoming Fax server with extensions using only one PSTN line, like this: PSTN number: 1234567, that's the line connected, FAX numbers like: 1234567-00, 1234567-01, 1234567-02 and so on are routed to their respective recipients. I do not want real DID (which would mean buying a bunch of numbers from the telco,

Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

<div><BR>Hello,</div> <div>&nbsp;I bought a TDM400P, and intended to use it with my analog phone, which is RJ11 ofcourse. So, the question now, how do I plug in my RJ11 phone to the TDM400P card, which has an RJ45 interface? Also, since it's an 11B card, I also intend to bring in an analog line into the RJ45, so i am still left with the same question....how do I go

Dear all, I am happy to tell you that I received a Digium's TDM20B card for my Asterisk box today. but the phone jack is RJ45 instead of RJ11. So Please I need precise instructions to connect a phone to this card. please, assume that I have a phone (a normal analouge phone connected to the one end of a cable with an RJ11 jack (at the phone side). and now I want to connect the other end to the

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

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, is there a way to limit the sip session per username?. i mean, if i have a sip session with asterisk using xxx as username, nobody can register with that username until my session is terminated. thanks

Hi, i have just migrated from courier-imap to dovecot. im using it with vpopmail, the pop service is working ok but the imap service for imp webmail is not writing the quota information to the maildirsize file, as recommended in the docs, i have this in dovecot.conf : plugin { # Quota plugin. Multiple backends are supported: # dirsize: Find and sum all the files found from mail