VoIP.brujula.net
Your Ad Here
english español brasil italia france germany china korea japan arabic russia

Ayuda y Manuales
  • Indice
  • FAQ
  • Beneficios
    SoftPhones
  • X Ten Lite
  • X Ten PRO
  • X-Lite V3 (with video)
  • EyeBeam
    Hardware Phones
  • Generic ATA
  • Sipura SPA 2000 2100
  • Sipura SPA 3000
  • Grandstream BudgeTone
  • LinkSys PAP2-NA
  • LinkSys RT31P2
  • LinkSys WRT54GP2-NA
  • LinkSys WIP300
  • Cisco ATA 186
  • Cisco IP Phone
  • Mitel Phone
  • Diphone D10 Plus
  • Dlink dvg1120 DPH 541
  • UTstarcom F1000 F3000
  • Uniden UIP 1868
    Mobile Phones
  • PalmOS
  • Nokia Series E, Series N
  • Nokia E61, E62 Nokia N70, N80
  • fring
  • iPhone
  • Windows Mobile 6
    VoIP IP PBX
  • 3CX Windows
  • Asterisk
  • TrixBox
  • @Home 1.5 @Home 2.7
  • SER Proxy
  • Cisco CallManager
  • Linux VoIP Server
    Comprar SIP Phones
  • Top Phones
  • Lista Completa
  • T-Shirt y Swag!
    Más Info VoIP
  • Codec FAQ
  • Test your Speed
  • Tutoriales VoIP
  • VoIP Videos
  • Noticias VoIP
  • English
    Preguntas ? voip@brujula.net
  • Asterisk PBX

    Configuration as regular SIP client

    In sip.conf under [general] add a register definition:
    
    Format:
     register => user:secret:authuser@host:port/extension
    
    
    Example (Register 2345 at sip provider as 1234):
     register => 2345:password@voip.brujula.net/1234 
    
      * user is the user id for this SIP server (ex 2345)
      * authuser is the optional authorization user for the SIP server
      * secret is the user's password
      * host is the domain or host name for the SIP server. This SIP 
        server needs a definition in a section of its own in SIP.conf 
        (mysipprovider.com).
      * port send the register request to this port at host. 
        Defaults to 5060
      * /1234 is the Asterisk extension that will be used for incoming 
        calls. 1234 is put into the contact header in the SIP Register 
        message. This is used by the remote SIP server when it needs 
        to send a call to Asterisk. If you want to process the incoming 
        call then 1234 must also be defined in extensions.conf 
    
    The server definition for outgoing calls looks like this:
    
     [voip.brujula.net-out]
     type=peer
     secret=password
     username=2345
     host=voip.brujula.net
     fromuser=2345
     fromdomain=voip.brujula.net
     nat=yes
    
    In extensions.conf you'd then use a statement like this:
    
     exten => _9.,1,Dial(SIP/${EXTEN:1}@voip.brujula.net-out,30,r)
    
    Please note that the ${EXTEN:1} variable here extracts all 
    but the first characters from the current extension (the 
    current match), in this case: 9 + the following digits. 
    Refer to the Asterisk variables Substrings section for 
    more details
    
    

    Configuration for AZ termination users
    (high volume users - requires ACL of your ip on our side)

    sip.conf

    ;
    ; SIP Configuration for Asterisk
    ;
    [general]
    ;disallow=gsm
    ;allow=ulaw
    port = 5060            ; Port to bind to
    bindaddr = 0.0.0.0     ; Address to bind to 
    context = from-sip     ; Default for incoming calls
    callerid=No CallID
    
    ;  This simply dumps calls at voip.brujula.net via SIP
    ;  There is no username/password required, since this 
    ;  is simply a SIP gateway, and not a proxy.  
    ;  Protection provided by ACLs on the router.
    ;
    ;
    
    [voip.brujula.net]
    context=brujula
    type=friend
    host=200.68.120.81
    
    
    
    Mi Cuenta VoIP.brujula.net
    Número SIP
    Contraseña:
     
    Olvido su contraseña?
    Crear cuenta SIP gratis!
    80.000 usuarios en 195 paises


    VoIP.brujula.net. We provide Internet phone service with free Internet calling and unlimited US, Canada, Europe and World plans. We offer prepaid phone service using our voice over IP system and an analog telephone adaptor. The solutions are designed for home phone service, business phone service, call shops and cyber cafes. VoIP.brujula.net supports Xten / Counterpath SIP softphones and Internet telephony equipment such as Sipura 2000, Sipura 3000, Cisco 186, Linksys PAP2, RT31P2, WIP300. D-Link DVG-1402SL, UTstarcom F3000. Nokia E60 61 N70 95. Symbian OS. Windows Mobile. Palm Treo, Pirelli Dualphone, Twintel, T-Mobile Dash. We also support Asterisk PBX and offer VoIP PBX software for businesses, resellers, ITSPs and campus applications.