similar to: RDSI vs Analogic

Hi. Can you give me some hints on how I can create a rotational board? I dont even know how to spell it in english. What I want is to have more than one line reserved, but with a single phone number, so that people can call to the same number and get a ringing signal if any of the lines is available, instead of having to dial 5 different numbers in order to get a free line. This is done

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

Rodolfo, I haven't looked up how to do this with sip phones, but the zap channels can be configured in groups that will hunt through the group until a non-busy line is found. http://www.voip-info.org/tiki-index.php?page=Asterisk%20ZAP%20channels#comments Here is a link to PBX hunting with the dial plan. http://www.voip-info.org/tiki-index.php?page=PBX+Hunt+Groups I haven't tried

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

Hey all! I had an experience trying to set up an E1 in Brazil which could help somebody. In Brazil is very common telcos to have just R2 digital as their primary signaling. As I were trying to set up an E100P, which does not support R2 yet, I had to test an other signaling which works perfectly with Asterisk. They call this signaling as RDSI, using ccs as framing and PA (primary access) as

Hello guys, I have this problem when a call is received in my PBX: (Caller) --> (Redirecting Service) --> (E1 PRI) --> (Asterisk PBX) --> (Internal Phone) Reception works fine, but when conversation finishes and the agent at internal phone hangs up, the call at caller's side is still alive for many seconds until it hangs up. The problem is that Telephone Company is billing me

Hi. I'm using asterisk as a PSTN -> SIP gateway, so that you can call to any of the 4 PSTN lines connected to the asterisk box from and dial your number, and asterisk will dial out through one of the 4 sip accounts I have on a SIP -> PSTN provider. I think of something like this in the extensions.conf [incoming] exten => s,1,Wait,1 ; Wait a second, just for

Hi all, We need help with our SuSe9.2 asterisk box We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone. We have downloaded the bristuff (0.2.0-RC7j) and installed it without problems. once we downloaded and compiled asterisk, zaptel and all other stuff, the module installation succed in this order: modprobe zaptel modprobe qozap modprobe wcfsx then the ztcfg output this: Zaptel

Hi all, this time with the complete configuration files... We need help with our SuSe9.2 asterisk box We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone. We have downloaded the bristuff (0.2.0-RC7j) and installed it without problems. once we downloaded and compiled asterisk, zaptel and all other stuff, the module installation succed in this order: modprobe zaptel modprobe qozap

Hi. I cant make SIP calls from asterisk. When I start asterisk, I get the following message: What does it means?? Asterisk is not behind NAT or Firewall. ---------------------------------- [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Sep 16 09:52:33 WARNING[16384]: chan_sip.c:8477 reload_config: Unable to get IP address for

Hi all, I'm newbe with asterisk and i'm facing with this problem that i'm not able to solve. I've to put an asterisk box after an analogic pbx wich require a 0 digit to give the dialtone. So when a client ask asterisk to dial an extension it should 1) send the 0 digit 2) wait for the dialtone 3) dial the extension the client send. How can i obtain this result? Thank's in

Hi, I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. Thanks. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0436-4, 03/09/2004 Tested on: 07/09/2004 13:57:36 avast! - copyright (c) 2000-2004 ALWIL Software.

Hi again. My last post was incomplete, so I' reposting it. I've seen that each country has its own PSTN qualities. I would like to know the minimal characteristics needed in PSTN to use Asterisk and also if some body knows which are Spain's PSTN's. I'm interested in buying a TDM card (probably a TDM11B card) and I need to know if it will fully work on Spanish PSTN and what

Hello all, I`ve got an AVM c2 card instaled on my SuSE box. I?m having problems configuring its channels. I don?t know how to set up asterisk to use the CAPI channels. I don?t know how to call them. My capi.conf is as follow, [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] ;#########config de la primera interface CAPI##########333

Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *********************************************************** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten

Hi there, I''ve read the paper on Xen live migration, and it shows some very impressive figures, like 165ms downtime on a running web server, and 50ms for a quake3 server. I installed CentOS 5 on 2 servers, each with 2x Xeon E5335 (quad-core), 2x Intel 80003ES2LAN Gb NICs. Then I installed 2 DomUs, also with CentOS 5. One NIC is connected to the LAN (on the same switch and VLAN), the

Hi to all rsync users. rsync's `--delete' option works fine in the following example: I'm sending all the content of /home/rodolfo from machine1 to /home/rodolfo in machine2: $ rsync --dry-run -vrtul --delete --exclude='/.*' . 192.168.0.2:/home/rodolfo , and --delete works perfectly. Instead, in this other example: $ rsync --dry-run -vrt --delete --modify-window=1 file1

Hi there, I got really tired of false hangups, specially when someone calls from SIP/IAX/whatever to PSTN, which makes no sense in using busydetect in the zap channel, as the caller will eventually hear the busy tones and hangup, causing the zap channel to be "freed" as well... After suggesting that the busy detect be enabled only for inbound calls, and finding other people that think