[DGrant303]

I have recently installed Asterisk@Home 2.8 and trying to configure Broadvoice as my carrier. I have signed up for an account with Broadvoice and have all my user info.

I guess I'm confused. Should I actually edit the sip.conf or sip_additional.conf files or if I used the FreePBX Administration is that sufficient enough?

I have tried everything and can't get it to work. When I try to dial out, I get the recording, "All circuits are busy..."

Does anyone have the setting information for this and how to edit. Also, everything I have read on the internet mention I need to edit the etc\host.conf file. I've tried editing this file and get a permissions error stating info no written. Am I missing something?

Any help would be greatly appreciated.

Thanks,

David

[Riceman]

If you post your config found in sip.conf and sip_additional.conf someone here should be able to help. Also is it behind NAT, what type etc.

Also make sure that in your outbound route you are set to use the sip trunks you created and not the zap/g0.


Later,


Nick

[DGrant303]

my 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.
register = 3033091083@sip.broadvoice.com:*****:...broadvoice.com/3033091083

[general]

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

My sip_additional.conf:
register=3033091083@sip.broadvoice.com:*****:30330 91083@sip.broadvoice.com/3033091083

[3033091083]
username=*****
user=phone
type=user
secret=*****
insecure=very
host=sip.broadvoice.com
fromuser=3033091083
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=yes
authname=3033091083

[4000]
username=4000
type=friend
secret=4000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=4000@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=David <4000>

How do I know if I'm behind NAT?

[Riceman]

If you are behind a linksys or anything like that you are probably behind NAT. If your address on the asterisk box is 192.168.x.x or 172.16.x.x you are behind NAT. If so you should have nat=yes in your sip.conf and on your sip trunks


Later,


Nick

[DGrant303]

where in sip.conf should I place the nat=yes and what do you mean by placing it in the sip trunks? Where?

[chandave]

nat=yes|no|rfc3851|route|never can be setup in the [general] section.

It can also be used when specifying peers (type=peer), users (type=users), or friends (type=friend).

The NAT setting informs Asterisk about the connection behavior of the either end.

When NAT=yes, Asterisk will send signalling information (i.e. the SIP message used for call setup, teardown, canceling, etc), back to the original source of the SIP message. It ignores any information placed into the SIP message's header about where to send things and where you appear to be when talkin with the destination.

When NAT=no or NAT=rfc3581, Asterisk will look a the SIP message headers and try to deduce what IP address and port Asterisk appears to be using to the Public Internet. It will then craft SIP message to others using that information.

When NAT=route, Asterisk will behave like NAT=rfc3581, but it will omit the "rport=" message header. The "rport=" indicates to the other end that you want them to not only fill in the information about your Public IP address but also the port your Asterisk appears to be using. With NAT=route, is not sent. This was to work around a bug with some old hardphone firmware (the SNOM Phones).

When NAT=never, Asterisk assumes that it is directly connected to the Internet with a Public IP address.

If you are behing a home router, most of the time, you will need to set NAT=yes if you are using an VoIP Service Provider.

For information about all the configuration parameters inside sip.conf, look at the Voip-info.org Wiki.

See ya...

d.c.

[DGrant303]

I am still having problems getting Broadvoice to work with Asterisk 2.8.

If anyone has any suggestions, it would be greatly appreciated. Below is my SIP.CONF, SIP_ADDITIONAL.CONF and what my debug report shows.

Also, from the debug report, it looks like it may be a connectiviity issue??? I have a DSL connection with an Actiontec modem. The IP address is 192.168.0.1. The DSL modem is conencted to a wired/wireless router and that IP address is 192.168.2.1. I have my IP on the Asterisk box to 192.168.2.5, the gateway and nameserver IP's are set to the DSL modem 192.168.0.1. Is this correct?

SIP.CONF FILE:

; 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]

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=3033091083@sip.broadvoice.com:********:30 33091083@sip.broadvoice.com/4000

[3033091083]
username=3033091083
user=phone
type=user
secret=*******
nat=yes
insecure=very
host=sip.broadvoice.com
fromuser=3033091083
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=no
authname=3033091083

[4000]
username=4000
type=friend
secret=4000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
mailbox=4000@device
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=David <4000>

[sip.broadvoice.com]
username=3033091083
user=phone
type=peer
secret=*******
qualify=yes
insecure=very
host=sip.broadvoice.com
fromuser=3033091083
fromdomain=sip.broadvoice.com
dtmfmode=inband
dtmf=inband
context=from-pstn
canreinvite=no
authname=3033091083

DEBUG REPORT:

May 11 07:57:41 WARNING[4347] chan_sip.c: No such host: sip.broadvoice.com
May 11 07:57:41 WARNING[4347] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
May 11 07:57:47 DEBUG[4570] manager.c: Manager received command 'Command'
May 11 07:57:47 DEBUG[4570] manager.c: Manager received command 'Command'
May 11 07:58:01 NOTICE[4347] chan_sip.c: -- Registration for '3033091083@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #25)
May 11 07:58:01 WARNING[4347] chan_sip.c: No such host: sip.broadvoice.com
May 11 07:58:01 WARNING[4347] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
May 11 07:58:01 WARNING[4347] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)


THANK YOU!

[chandave]

From the linux command line, can you issue the command:

host sip.broadvoice.com

If it returns:
Host sip.broadvoice.com not found: 3(NXDOMAIN)

then you probably have connectivity issues. Check and make sure you can access the Public Internet from your Asterisk box.

See ya...

d.c.

[DGrant303]

I get the following message:

sip.broadvoice.com is an alias for proxy.chi.broadvoice.com. proxy.chi.broadvoice.com has and address of 147.135.12.128

I looked at the debug log and now this is what it's showing:

May 11 08:39:43 WARNING[2800] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
May 11 08:40:03 NOTICE[2800] chan_sip.c: -- Registration for '3033091083@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #19)
May 11 08:40:03 WARNING[2800] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)
May 11 08:40:03 WARNING[2800] chan_sip.c: Probably a DNS error for registration to 3033091083@sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds)

[dswartz]

btw, since you said "asterisk 2.8", i'm guessing you're using asterisk@home 2.8? this is not the same as asterisk 2.8 (which doesn't exist - the latest regular asterisk is 1.2.7.1). asterisk@home is regular asterisk with a bunch of addons. that said, if you ARE using AAH, you should NOT be editing .conf files directly (except if they have "_custom" in the pathname, as otherwise AAH will overwrite your changes...