***SOLVED*** [Asterisk-Users] Broadvoice latest changesandstillnot working- An Additional Server****Solved*****!

[Zanzamar Majere]

I had the same problem.  But I had a Sip 400 Bad Result  ...   Failed to 
authenticate on INVITE ...

I am running asterisk 1.03

I do not have my program pointing to any proxies...It is pointing to 
Sip.broadvoice.com, I do not have any proxies set up in my /etc/hosts file
where PPPPPPPPPP = Phone Number
XXXXXXXXX = secret

Try cutting and pasting mine in, see if it works...

My Sip.conf is as follows:
[general]
context=sip                     ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port is 
5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
dtmfmode=inband
disallow=all
allow=ulaw
allow=gsm

register => PPPPPPPPPP:XXXXXXXXXX at sip.broadvoice.com

[PPPPPPPPPP]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=PPPPPPPPPP
secret=XXXXXXXXXX
username=PPPPPPPPPP
insecure=very
context=sip
authname=PPPPPPPPPP
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no



On Thursday 10 March 2005 02:08 pm, Joe wrote:
> Zanzamar,
>
>  I agree that it should work.  I can call out and have the land phone
> ring,  but as soon as it is answered, another invite goes out and that
> is when I get the 401 not authorized. I don't want to go down this
> route,  but could this be a Codec issue?
>
> Here is my sip config
>
>
>
> [sip.broadvoice.com]
> type=peer
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser= BBBBBBBBBB
> username= BBBBBBBBBB
> authuser= BBBBBBBBBB
> secret= secret
> context=sip
> nat=no
> insecure=very
> dtmfmode=inband
>
>
>