Broadvoice - Inbound Calls


#1

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


#2

what does your SIP.CONF and SIP_ADDITIONAL.CONF look like?


#3

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


#4

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


#5

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.


#6

I dropped the last 2 digits for privacy.


#7

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


#8

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