search for: rodolfograv

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

...fo.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 this myself, but I plan to soon. Ty Purcell -----Original Message----- From: Rodolfo Grave [mailto:rodolfograve@yahoo.es] Sent: Monday, September 20, 2004 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [QUAR][Asterisk-Users] How can I make a rotative board? 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. Wh...

Hi. I'm getting new lines for using with Asterisk. In my Telco they said I could choose between Analogic lines and RDSI lines... I've already bought a TDM400P with FXO modules. Can you give some hints on the differences between RDSI and normal Analogic lines? Would I have problems for using a RDSI line with the TDM? Any other issue in general? Thanks in advance, RODOLFO --- avast!

...... please HELP (Kristian Kielhofner) 12. RE: polycom and cdp (Richard) 13. Re: RE: Zaptel/Zapata config from T410p to BrooktroutT1 (jbebeau) ---------------------------------------------------------------------- Message: 1 Date: Thu, 23 Dec 2004 03:32:22 +0100 From: Rodolfo Grave <rodolfograve@yahoo.es> Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please HELP To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <41CA2E36.9030802@yahoo.es> Content-Type: text/plain; charset=ISO-8859-1; format=flowed...

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

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

Hi. I've succesfuly installed Asterisk (at least I think so since compilations were cool). I havent installed my x100p yet and I wish to make a call from the command line to test my configuration.... is it possible? I've seen there is a "sample.call" file at the asterisk source dir, it says you can place a call by dumping it into "/var/spool/asterisk/outoging". Does

Hi. I have a x100p card installed and also asterisk, but I just dont get asterisk to register with my sip provider (FWD)... when I start asterisk using the following command I get the following messages (first, a lot of messages show up immediatly after starting up: I'read this is normal, then the CLI console comes out and this messages appear): NOTICE[229390]: chan_sip.c:3922

Hello. I'm having this problem with a Budgestream phone: I've correctly installed G729 licensed codec in my asterisk box, but when I set my budgestream to use only g729 codec, asterisk throws this message: *CLI> Dec 21 18:02:49 WARNING[2375]: chan_sip.c:2764 process_sdp: No compatible codecs! Dec 21 18:02:50 NOTICE[2375]: chan_sip.c:7295 handle_request: Unable to create/find

Hi all. I've installed a TDM card, with 1 FXO port. I've configured the zaptel driver and everything seems to be ok: Asterisk answers the calls. Now, the problem is that even when the caller hangs up ("the caller" is my self from another PSTN line) Asterisk doesn't detect it, and it goes on processing the current context in extensions.conf as if the call was still

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hello... I'm having problems with H323/G729 setup. Below is the output of h.323 debug when making a call. I use a SIP phone connected to an * box in the same LAN. The * connects to a h323/g729 PSTN terminator through internet. Calls rings and are answered in the other side, but I get no sound at all nor the other side does (complete silence in both sides). I thought this would just happen

Hi. I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm going to install: 1-)One X100P (1 FXO module) 2-)One TDM03B (3 FXO modules) I'll have the 4 FXO channels busy almost all the time, and I would like quality to be as good as possible without going to the high-level hardware. I would like to learn of some tested configurations (I've heard of problems

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

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 again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

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. 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