[Asterisk-Users] X101P CallOut Big Problem.

Hi all, I just now receive the FXO X101P Card but can't at any way make then call out. I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers. I play alot with txgain and rxgain, but none help me out. Being honest i

[Asterisk-Users] X100P CallOut Problems !!

Hi all, I just now receive the FXO X101P Card but can't at any way make then call out. I can hear the signal, even call but always receive from my local operator error that or the number don't exist or need more numbers. I play alot with txgain and rxgain, but none help me out. Being

[Asterisk-Users] About HT-286

Hi, Did HT-286 a good answer to put into my company and with both * and my old pbx use VOip and Normal telephones without change any kind of structure ? Like with HT-286 can i just plug it using the RJ-45 part in the * and the RJ11 in my old pbx and have both working with my normal telephone ?

[Asterisk-Users] Grandstream and HT-286

Hi, Did GrandStream Voip-phone and HT-286 Analog adapter talk using GSM 6.01 Codec ?? In tests using asterisk this codec it's the best for my kind of connection, did both Hardwares use this codec to talk ? In page they don't mention this. Thanks alot. Ps. ILBC ?? Talk too ?? Thanks

[Asterisk-Users] Sipura 2000 SPA-2000, Help Question.

Hi, Can anyone tell me if SPA-2000 sipura, talk GSM and bypass from a normal PBX and Asterisk to a analog phone ? I want to use SPA-2000 Adapter in my office with both my * and the old PBX that has in it. Can sippura byppass the calls from both to my analog phone ? So i can receive calls from

[Asterisk-Users] HT 286 Any information about will be great !!!

Hi, Did HT-286 Bypass calls from normal PBX and Asterisk PBX to analog phones ? To be more precisely, can i receive both call from this two kind of tecnologies using HT-286 in my office ? I dont want change my OLD PBX (That works great) with Asterisk and lose investiment etc. So i think in use

[Asterisk-Users] Best VOIP Analog adapter ???

Hi, Did anyone know if exist some adapter that give me the option to connect two kind of tecnologies ? Something like with 1 RJ-45 port 1 RJ 11 Port (IN), and 1 RJ 11 port (OUT). Then i can join my old PBX that works perfectly with Asterisk that works great too (But in voip mode) with my analog

[Asterisk-Users] Two FXO Cards answering at different times.

Hi all, Anyone know how put my X101P cards to answer at different ring times ? Like x101P(a) Answer at 3 rings x101p(b) Answer at 4 rings My * it's connected into a PBX thats when receive a call send to two lines at same times a ring. (So i must have a way to just put one channel to answer

[Asterisk-Users] Zaptel problems with Silence Detection

Hi people, There is any way to control silence detection in Zaptel ?? I have a x101p card, and sometimes the sounds dont come. I notice that this is a Silence Detection in the card how can i avoid this ? Another things, g.729a has silence detection ? Or Asterisk do this ??? Using a sip i can

[Asterisk-Users] Ebay X101P Card. CRAP!!!

Hi People, I know that this is a Digium forum, and actually i will buy cards now from Digium too. But a have just a question. For test purposes and of course save some money a buy from Ebay a " Mercury M/N: AMI-IA92 card." With this card Asterisk work well - my linux appear like "Tiger Jet card".

[Asterisk-Users] Asterisk and ISDN

Hi, I'm trying to use Asterisk with one ISDN TELES 16.03 c PnP Card (ISA) Now i can call Asterisk with the Modem i4l driver etc But need more information to make a better config and also know how call this card to make outgoing calls and receive incoming calls. I know how use with a Zap card, but

Re: [Asterisk-Users] Howto: Installing Asterisk and ISDN on Fedora Core 1

ISDN Carlos Arnt [EMAIL PROTECTED] Diretor de Informática. Divisão de Tecnologia e Desenvolvimento TI. Intellissence do Brasil. http://www.intellissence.com/brasil Tel:(+55)-(21)-(3908-4667) Tel(Direto):(+55)-(21)-(3905-1561) Cel:(+55)-(21)-(9169-8537) --VoIP Contact Method-- World VoIP Pin/Code

[Asterisk-Users] Best VOIP PSTN Provider.

Hi People, I wondering here, who is the best VOIP PSTN Provider to use with my * box ? That has good prices, good quality (Use ex. G729 codecs) etc ? I want to make calls to Europe, Asia etc with cheap prices and good quality in the sound ;) Did anyone has a good tip for me ? For use with IAX2

[Asterisk-Users] VMAIL.cgi

) etc etc Thanks Alot ! Carlos Arnt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] DTMF x-lite

Well now it's two with this problem. I try everything but don't work until now.. try chmod +s try chmod a+s etc etc etc ... Try 4755 etc. None works.. Put the /var/spool/asterisk/vm with apache.apache AND nothing.. So if anyone has a tip, please help. Another question is, did asterisk have any

RE: [Asterisk-Users] DTMF x-lite

. From: [EMAIL PROTECTED] [mailto:asterisk-users-[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Friday, October 31, 2003 5:45 AM To: [EMAIL PROTECTED]; asterisk-[EMAIL PROTECTED] Subject: RE: [Asterisk-Users] DTMF x-lite Well now it's two with this problem. I try everything but don't work until now

Re: [Asterisk-Users] Asterisk prepaid debug

What kind of prepaid agi did you use ?Could you send me the page ? How install or where find ? Thanks alot. On Mon, 19 Apr 2004 20:07:14 -0600, Julio wrote: My Asterisk prepaid debug is: - Hungup 'Zap/2-1' Urgent handler -- Starting simple switch on 'Zap/2-1' Urgent handler -- Playing

[Asterisk-Users] Grab phone call ?

Hi Let's say i have a call to a extension 115. But i'm under the extension 118 how take the call from 115 to my extension using * ?? Thanks alot. ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Zaptel Hangup always..

Hi all, Did anyone see this problem. I have two X101P in my asterisk Box. When i receive a call, everything goes Ok, but after a short period of time i receive a hangup .. Sometimes happen at 1:50min others at 05:00min but always happend. Did anyone see this before ?? How can i make the cards use

[Asterisk-Users] VOIP CBQ BandLimit HELP!!

Hi all, Reading about CBQ on internet i can say "I dont understand well" ;) So anyone that has a good background can help me out with this simple question ? I just want priorize my UDP packets to always has 90% of my link when use a VOIP connection with asterisk. My asterisk run in the same

Re: [Asterisk-Users] VOIP CBQ BandLimit HELP!!

That's great. Maybe i will ask a nonsense question. Let go then : Sip uses RTP right ? So open a SIP channel 5060 i have alot of RTP packets . Did i don't need mark this RTP packets too ?? I mean IAX2 use RTP ? In you script i see that i MARK IAX2 then i can control the rate and give to my VOIP

[Asterisk-Users] Dial 0 to outbound

Hi Folks. I see that can put 0 to call out using a x101p (zaptel) or even a pstn service. Thats great, but when press the 0 i just dial then the numbers to call out. There is any way after hit 0 (ear) the line sound ?? I know it's just a style way put some users, really like it !! So after hit 0

Re: [Asterisk-Users] New IAX software phone (for WIndows platform)

Hi Dan, It's a great program, Just a question, it's open source right ? Can i see the code ? I'm a c++ programmer too with some time to spend now (On vacation) :) So can i help ? Did you plan put in the page a source code for people download too ? About the ACTIVEX idea it's great too !!

[Asterisk-Users] Xphone Beta 1.01 ?

A simple question. I found an old Xten Xphone Beta 1.01 that has Sip capabilities. So i try with my asterisk, but he always try to login using this format: sip:[EMAIL PROTECTED]:5060 And say that registration failed. Someone see this kind of thing before ? I try using Xten Lite and everything

[Asterisk-Users] Asterisk over VPN.

Hi People, Let's take a look in this diagram : Part A - Server running VPN IP ie.192.168.10.1 Part B - Client running over the VPN with internal IP ie. 192.168.10.2 -- From network A i can reach B. Use all programs - Share Printers , aplications, using Netmeeting etc.. Then i make this in the

Re: [Asterisk-Users] DIAX version 0.9.2 available for download

a registry stuff, I think he's asking you to put the same option into DIAX, so people can use: callto: tel: or maybe : diax://192.168.x.x and open the diax dialing directly the specific ip or address. If i'm wrong in my tip sorry for that :) Just trying to help. Carlos Arnt []'sOn Tue, 11 Nov

Re: [Asterisk-Users] DIAX version 0.9.2 available for download

My explanation it's ok. Dan, it's great what he think in use. So into a groupware app he can put into ex. the name of the person a link to call then over diax too. But to be honest your ActiveX version will be more helpfull than just using a link. hmm, but it's a good idea too . Carlos []'sOn

Re: [Asterisk-Users] DIAX version 0.9.2 available for download

Well that's easy then . Look here : Windows Registry Editor Version 5.00 [HKEY_CLASSES_ROOT\callto] @="URL: DiaxTo Protocol" "EditFlags"=hex:02,00,00,00 "URL Protocol"="" [HKEY_CLASSES_ROOT\Diaxto\DefaultIcon] @="\"C:\\Program Files\\Diax\\diax.exe\",1" [HKEY_CLASSES_ROOT\diaxto\shell]

Re: [Asterisk-Users] DIAX version 0.9.2 available for download

Yep, that's right. On Tue, 11 Nov 2003 15:06:19 +0200, Dan wrote: Hi Carlos, - Original Message - From: Carlos Arnt To: [EMAIL PROTECTED] Sent: Tuesday, November 11, 2003 2:36 PM Subject: Re: [Asterisk-Users] DIAX version 0.9.2 available for download So into a groupware app he can put

[Asterisk-Users] (ast_rtp_read): Unknown RTP codec 72 received

Hi, Please someone know what can be this message ? "(ast_rtp_read): Unknown RTP codec 72 received" Always i try to use the X-lite, this message appears. What is the 72 number ? what means ??? I use sometime and never see this, now from nothing just appears.. Thanks for helping. Carlos.

[Asterisk-Users] Group dial codes ?(Newbie question)

Hi All, Using asterisk and extension.conf can i make a group dial code ? Like this. Ie. Let's say i have a group called directors. Only People in this group can dial to a external number like 800. How can i make this possible in asterisk ? Thanks alot ! Carlos.

[Asterisk-Users] Asterisk and voice recog support ?

Hi, Calling the FWD, i see a feature a little different. I don't call any number, but TALK with the system and they go to others parts of the showed menu. There are any way to make the same with * ? Where are the link that i'm talking about. http://fwd.pulver.com/callme.php?userid=5

[Asterisk-Users] MSN 4.7 and Asterisk.

Hi All, I'm just testing * with MSN 4.7 and works great, but when i try to call using the dialpad of MSN all my number into * appers twice. Like 100 when i try appears 1 .. Like a echo in the number. I'm using rfc2833 because inband just crash ... sip.conf. [msn] type=friend host=dynamic

[Asterisk-Users] MSN MESSENGER 4.7 with Asterisk -SOMEONE HELP HERE PLEASE!-

Hi all, I just trying to test MSN 4.7 that has SIP. Because with him i can use a video and voice transmission and * . But when i try to call someone using the DIALPAD of MSN, when i insert any digit into * the numbers appears twice !! like this. channel 456 appears in asterisk 445566 How can

[Asterisk-Users] Best SIP PHones to buy ?

Hi People, Can anyone help-me here with a simple question. I wanna buy a Sip Phone, but what is the best and cheap one ? I see alot of messages about, grandstream , snow etc etc. So for use with my * system, what sip phone is the best ?? Can him be used behind a nat system etc ? Or with a

Re: [Asterisk-Users] First version of the ActiveX version of DIAX (0.1.0) available for download

I put like the readme.txt say the code in my web page , put the OCX in the same Directory, but not work. Did has any problem ?? Thanks The code: html head meta http-equiv="Content-Type" content="text/html; charset=windows-1252" titleNew Page 1/title /head body OBJECT ID="diax" CLASSID=""

[Asterisk-Users] SpeeX and IAX2 ??? Work or not ??

All progs that i see only use GSM codec format to IAX2. What program can i use to test Speex Codec after all ? What i need put into iax.conf to force just use this codec ? "dissallow = all" " Allow = SpeeX " ??? I try IaxComm thinking that he uses Speex, but only call GSM too. Xlite can't use

Re: [Asterisk-Users] SpeeX and IAX2 ??? Work or not ??

Yep I know that. I just say X-lite because even in Sip can't work. But I see alot of people saying that use IAX2 with SpeeX codec well. I just be one of then too :) On Mon, 29 Dec 2003 08:34:32 -0500, Andrew Thompson wrote: - Original Message - From: "Carlos Arnt" [EMAIL

Re: [Asterisk-Users] Fax

IntelliFAX Como funciona. Sua rede fica ligada a internet bem como seus fornecedores. Você passa o fax normalmente que se encotra ligado em nosso PABX Virtual e seus fornecedores via Internet usando uma versão light do sistema poderão receber de qualquer canto do mundo este fax em seus faxes.

Re: [Asterisk-Users] Fax

Só que existe um problema ! O sistema deles * não funciona com o seu !! O seu e desenvolvido em Windows (Delphi) enquanto que o deles e Linux !!! O * e totalmente VOIP o seu no momento fala um protocolo louco !! Me liga então. Ps. Ao invês de usar sua conta e o forum quer por favor, usar e-mail

[Asterisk-Users] Asterisk Compilation with H323 working on it

Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323 WORKING ? I try use H323 with Asterisk for some implementations but that cant good results. So any tip ? Thanks alot ! Carlos. ___ --Bandwidth and Colocation

RE: [Asterisk-Users] Asterisk Compilation with H323 working on it

] [mailto:asterisk-[EMAIL PROTECTED] On Behalf Of Carlos Arnt Sent: Thursday, October 20, 2005 2:24 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Compilation with H323 working on it Hi Folks, Can recomend a asterisk compilation for Mandrake or Debian that has on it H323

[Asterisk-Users] Routes IPSEc And Asterisk.

Hi Folks, Really need a help here. I have 3 network places Let´s say : A , B and C Using Freeswan (Ipsec) i make the point A see and interact with point B (Now both networks see each other) Everything is perfect, but i have in point B now a C Network that comes over Router. Point B com see

[Asterisk-Users] RTP Packets going over caller and calle !!

Hi, I Have a problem here, if anyone know a method to avoid please tell me . Using * with the option canreinvite=yes i can in theory tell to my * box, send RTP Packet directly from one Sip device to another one, then "In Theory", i will not use my own internet connection. So this mean that

[Asterisk-Users] Canreinvite=???

Hi, everyone ! Looking at this explanation : When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another