Asterisk + VOSP account working configuration?

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Asterisk Users 3 Comments

On Tue, 14 Dec 2010 16:56:14 +0100, Gilles
wrote:
>PS: Here’s what I’m thinking of using:

At this point, Asterisk seems to register OK with my VOSP, but when I
call the number from my cellphone, I get this error:

“NOTICE[88]: chan_sip.c:14033 handle_request_invite: Call from
‘myvospaccount’ to extension ‘s’ rejected because extension not
found.”

Incidently, how does Asterisk know how to link calls from the VOSP to
an extension in the dialplan?

Here’s what I’m using:

;================ sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0

;deny=0.0.0.0/0
;permit=
externip=
localnet=192.168.0.0/24
nat=yes

;all RTP packets go through Asterisk
canreinvite=no

disallow=all
allow=ulaw
allow=alaw
allow=gsm

;incoming calls from VOSP
;can’t use “s” extension?
context = vosp-incoming
register => myvospaccount:mypasswd@myvosp.com

;================ extension.conf
[general]
static=yes
writeprotect=yes
clearglobalvars=no
autofallthrough=yes

[vosp-incoming]
exten => s,1,Dial(SIP/6011)
exten => s,n,Hangup

Thank you.

3 thoughts on - Asterisk + VOSP account working configuration?

  • Gilles wrote:
    You are setting up a SIP trunk from your VOSP provider(whatever VOSP
    is). It dials your phone number. So whatever you dial from your cell
    phone is the extension that this trunk should land at.

    ‘s’ is not an extension. It’s a placeholder for the steps in your dial plan.

    For instance if my phone number with my provider is 815 555 1212, then I
    need an extension 8155551212.

    I would use:

    [inbound]

    exten => 8155551212,1,answer
    exten => 81555551212,2,Goto(mainmenu,s,1)
    exten => 8155551212,3,hangup

    Lyle Giese
    LCR Computer Services, Inc.

  • Le 15/12/2010 15:21, Gilles a √©crit :

    Why 2 context? Todays Asterisk versions only needs one peer context for
    incoming/outgoing. Something like

    [vosp]
    type=peer
    host=myvosp.com
    username=myaccount
    secret=mypasswd
    fromuser=myaccount
    fromdomain=myvosp.com
    nat=yes
    canreinvite=no
    context=from_vosp

  • On Thu, 16 Dec 2010 10:06:35 +0100, Administrator TOOTAI
    wrote:

    I tried combining the two sections in sip.conf, but get a BUSY signal
    for incoming calls from the PSTN. Could it be because I’m running
    Asterisk 1.4?

    http://www.ippi.fr/index.php?page=sip_parameter
    (scroll down to “PBX Asterisk – TrixBox”)

    Also, it’s important to have the Outgoing part come before the
    Incoming. Otherwise, I also get a BUSY signal for incoming calls.

    BTW, why do we need to define the login/password twice, once with
    “register” and a second time in a [_outgoing] section?

    Thank you.