[Asterisk-Users] BroadVoice - "Failed to authenticate on INVITE" error

[Howard Waterfall]

I'm experiencing a "Failed to authenticate on INVITE" error (see
output below) whenever I try to MAKE a call through the Broadvoice
account.  I noticed some others had the same problem but it went away
when they rebuilt Asteris w/ a new version.  N such luck for me!

I'd be grateful for any assitance.  Here's what I've done so far:

1) I downloaded the latest stable version of Asterisk and compiled it
(27-Mar-05).
2) I updated my conf files as per the Broadvoice web site (see below)
3) I CAN make and recive calls through the Broadvoice account using X-Lite.
4) To avoid typos, I used cut and paste in sip.conf to copy the phone
number and password from the register line  to the
[sip.broadvoice.com] section
5) When I run Asterisk:
   i)   The Broadvoice account registers OK
   ii)  I can receive calls on the Broadvoice account
   iii) I CANNOT make calls through the Broadvoice account.  When I
do, my computer freezes up but eventually comes around a while after I
hangup and warns - Failed to authenticate on INVITE to '"asterisk"
<sip:8145551212 at sip.broadvoice.com>;tag=as4a325b3a' (see below)

Any ideas?

Finally, I'm still unclear about assigning an extension to the
Broadvoice account as part of the  registration line (see where I
commented out ;/3003).  What does it do?  I rely on the context
defined under the [sip.broadvoice.com] section.  What do you gain by
assigning an extension in the Register line?

My conf files and the Asterisk output are below.

Thanks,
Jewel


;*****************************************************************
;/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
; 
; This is the bare bones of what is required to get your X100P 
;  card working on a "normal" line provided by a local phone
;  carrier in North America.   For more details on all options, 
;  see /usr/src/asterisk/configs/zapata.conf.sample but I would
;  strongly suggest starting simple with the bare minimum of
;  configs and working up from there - PSTN telephony interfaces
;  are notoriously touchy with the large number of features 
;  they offer.
;  
[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'