[Asterisk-Users] broadvoice

[hugolivude]

Woops forgot to include my config files...

;*****************************************************************
;/etc/hosts
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1       localhost.localdomain   localhost
# proxy.dca.broadvoice.com
147.135.0.128   sip.broadvoice.com
;
;*****************************************************************
;
;/etc/asterisk/sip.conf
;
[general]
port=5060                 ; Port to bind to (SIP is 5060)
bindaddr=0.0.0.0          ; Address to bind to (all addresses on machine)
context=from-sip-external ; Send unknown SIP callers to this context
pedantic=no
register => 8145551212 at sip.broadvoice.com:<password>:8145551212 at sip.broadvoice.com;/3003
;
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8145551212
secret=<password>
username=8145551212
insecure=very
context=from-broadvoice
authname=8145551212
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
;
;*****************************************************************
;
/etc/asterisk/extensions.conf
[general]
static=yes       ; These two lines prevent the command-line interface
writeprotect=yes ; from overwriting the config file. Leave them here.
;
;
[from-broadvoice]
exten => s,1,Dial(ZAP/1,30)
exten => s,2,Hangup

[from_FXS]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
;
;*****************************************************************
;
;/etc/asterisk/zapata.conf
;
;
[channels]
language=en
context=from-FXO
signalling=fxs_ks
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
channel => 4
;
language=en
context=from_FXS
signalling=fxo_ks
channel=>1
;
language=en
context=from-ILS-FXS
signalling=fxo_ksFailed to authenticate on INVITE to '"asterisk"
<sip:8145551212 at sip.broadvoice.com>;tag=as4a325b3a'
channel=>2
;
;*****************************************************************
;
;/Asterisk Console Output
;
Asterisk Ready.
*CLI> sip show registry
Host                  Username     Refresh State
147.135.0.128:5060    8145551212       120 Registered
*CLI>     -- Starting simple switch on 'Zap/1-1'
   -- Executing Dial("Zap/1-1",
"SIP/13035551212 at sip.broadvoice.com|30") in new stack
   -- Called 13035551212 at sip.broadvoice.com
Mar 27 20:55:26 NOTICE[1116941248]: chan_sip.c:5047 handle_response:
Failed to authenticate on INVITE to '"asterisk"
<sip:8145551212 at sip.broadvoice.com>;tag=as4a325b3a'
Mar 27 20:55:26 WARNING[1209214528]: app_dial.c:347 wait_for_answer:
Unable to forward voice
 == Spawn extension (from_FXS, 13035551212, 1) exited non-zero on 'Zap/1-1'
   -- Hungup 'Zap/1-1'

On Apr 4, 2005 2:29 PM, Matt <mhoppes at gmail.com> wrote:
> Hi,
> I'm currently routing my asterisk server out over broadvoice.. it
> seems I can do multiple outgoing and incoming calls.... does anyone
> know if broadvoice actually allows this or not?
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>