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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


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FonoSIP.com supports Xten / Counterpath SIP softphones for Windows, Mac, iPhone and Linux. Internet telephony equipment such as Linksys PAP2, Cisco, Sipura, Nokia N71. Apple iPhone, iPod Touch, iPad. Android. Codec G729. Fring Mobile. Bring your Own Device BYOD. We also support SIP Trunking, replace your phone lines. Asterisk PBX, TrixBox.