I have 3 trunks from Broadvoice. All outgoing calls work fine from each extension.
I can not get incoming calls no matter what I try.
Also I can not record messages.
Thanks
what does your SIP.CONF and SIP_ADDITIONAL.CONF look like?
AS I said I can make outbnoud calls and I get all numbers registered
Thanks
sip.conf
; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.
[general]
pedantic=no
externip=X.X.X.X
localnet=10.100.X.0/255.255.255.0
progressinband=yes
bindport=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)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
#include additional_a2billing_sip.conf
sip_additional.conf
register=95482714@sip.broadvoice.com:password:954827140@sip.broadvoice.com
register=95482714@sip.broadvoice.com:password:95482714@sip.broadvoice.com
register=95482715@sip.broadvoice.com:password:95482715@sip.broadvoice.com
[2000]
username=2000
type=friend
secret=2000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=2000@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Barry Kripitzer <2000>
[2005]
username=2005
type=friend
secret=2005
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=2005@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Carl Gren <2005>
[6000]
username=6000
type=friend
secret=6000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=6000@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Laura Kripitzer <6000>
[95482714]
username=95482714
user=95482714
type=user
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
context=from-pstn
[95482714]
username=95482714
user=95482714
type=user
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
context=from-pstn
[95482715]
username=95482715
user=95482715
type=user
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
dtmfmode=rfc2833
dtmf=rfc2833
context=from-pstn
callerid=95482715
[sip1.broadvoice.com]
username=95482714
user=phone
type=peer
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=95482714
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
canreinvite=no
authname=95482714
[sip2.broadvoice.com]
username=95482714
user=phone
type=peer
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=95482714
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
canreinvite=no
authname=95482714
[sip3.broadvoice.com]
username=95482715
user=phone
type=peer
secret=
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=95482715
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
canreinvite=no
authname=95482715
sip_additional.conf
I am having the same problem with vonage sip trunks, I bet if you set nat=yes in your sip.conf inbound will start working but outbound will fail. I am behind a linksys right now but have a monowall coming so hopefully it wll fix my problem. Here is a link to my thread: forums.digium.com/viewtopic.php?t=6126
I don’t claim to be an expert on these things, but it looks like your phone numbers are all wrong … BV wants the full 10 digit phone number on the registration and for the various SIP parameters (username, etc.). Your numbers look like you just have a 7 digit number prefixed by 9 everywhere (except where you added a 0 to the end of one of them) …
You also have two contexts with the same name and two registrations with the same number (as well as having the wrong numbers, I think). I doubt that * will do what you intend with those.
I think you need to fix a whole pile of things in here.
I dropped the last 2 digits for privacy.
Just had this problem and here’s what solved it for me:
sip.conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
pedantic=no
register => <phone #>@sip.broadvoice.com::<phone #>@sip.broadvoice.com
[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=<phone #>
secret=
username=<phone #>
insecure=very
context=default
authname=<phone #>
dtmfmode=inband
dtmf=inband
canreinvite=no
extensions.conf
[default]
exten => <phone #>,1,Answer()
exten => <phone #>,2,Playback(hello-world)
exten => <phone #>,3,Hangup()
Which plays back the hello world file. Not very useful, but at least it answered and incoming call finally (after a couple hours worth of tinkering).
I found this using the asterisk console command
sip debug peer sip.broadvoice.com
which displayed a lot of info, of interest was the line
Looking for <phone #> in default (domain )
Anyway, hope it helps
at the cli run sip debug make inbound call
look for “context” errors
is this just asterisk or do you have freepbx the gui???
if so go to beta 3 beta 1 and 2 have problems with inbound trunks
if beta 2 leave the cid and inbound did blank to test inbound