[Asterisk-Users] Asterisk With Broadvoice

[Christopher McBee]

Here is a copy of my config that works great with broadvoice.  I also
have an AGI that I wrote to verify country codes so your users can't
call countries that aren't included in broadvoices plan.  If you want
that too, just let me know.


Sip.conf
-----------------------------------------------------------------
; Inbound broadvoice calls
register => 8029041486:mypass at sip.broadvoice.com/8029041486


[Broadvoice]
type=friend
username=8029041486
fromuser=8029041486
secret=zjfg9f18fh
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
port=5060
dtmfmode=inband
insecure=very
permit=147.135.0.128/32
qualify=yes
canreinvite=yes
nat=no
----------------------------------------------------------------

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk at billwho.com
Sent: Friday, February 25, 2005 9:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice

OK,

After checking into this, I have found the following:

I can set it up so either incoming or outgoing sip calls on this trunk 
work but NOT both.  The "sip show registry" command shows everything as 
it should be.

The section from my sip.conf is as follows:

[Broadvoice]
username = 2xxxxxxxxx
type=peer
secret=password
nat=yes
host=sip.broadvoice.com
fromuser=2xxxxxxxxxx
fromdomain=sip.broadvoice.com
dtmfmode=inband
canreinvite=no

My registry string is:
21xxxxxxxx at sip.broadvoice.com:password at sip.broadvoice.com

If I remove type=peer from [Broadvoice] in sip.conf incoming calls work 
great but outgoing calls don't work.  If i leave type=peer in there, 
outgoing calls work great but incoming calls get routed to Broadvoice's 
Voicemail . . .


Roger Hanson wrote:

>
> ----- Original Message ----- From: <asterisk at billwho.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, February 24, 2005 10:12 PM
> Subject: [Asterisk-Users] Asterisk With Broadvoice
>
>
>> I have configured asterisk with the AMP php configuration utility.  I

>> am able to make outgoing calls through broadvoice but incoming calls 
>> are sent to BV's Voicemail and never actually enter the IVR.  When I 
>> show sip debug info through the asterisk prompt it actually reads the

>> incoming call from BV but then issues a busy signal sending the call 
>> to BV's voicemail.
>>
>> I also modified extensions.conf as follows:
>> [from-sip-external]
>> include => from-pstn
>>
>> I have set up my sip trunk in AMP as follows:
>>
>> Trunk Name: Broadvoice
>> Peer Details:
>> dtmfmode=inband
>> fromdomain=sip.broadvoice.com
>> fromuser=21xxxxxxxx
>> host=sip.broadvoice.com
>> qualify=yes
>> secret=password
>> type=peer
>> username=21xxxxxxxx
>>
>> My Incoming Settings are:
>> User Context: sip.broadvoice.com
>> User Details:
>> context=from-pstn
>> dtmfmode=inband
>> fromdomain=sip.broadvoice.com
>> host=sip.broadvoice.com
>> nat=yes
>> secret=password
>> user=21xxxxxxxx
>> username=21xxxxxxxx
>>
>> My register string:
>> 21xxxxxxxx at sip.broadvoice.com:password at sip.broadvoice.com
>>
>>
>
> Something to double check and something to try (in that order):
>
> 1.  check your password.  It's not the password you registered at 
> their website with.  They send you an email with a different password 
> in it you need to use.  The password you registered at their website 
> is just for logging into their website.
>
> 2.  Try using a standard registration string - not the one they show 
> you.  Use number:password at sip.broadvoice.com instead of the one they 
> show you on the website.
>
> See if one of those things is the trouble.
>
> If that doesn't work, look at "sip show registry" and see what's 
> registered.
> asterisk*CLI> sip show registry
> Host                                          Username        Refresh 
> State
> sip.broadvoice.com:5060         952225xxxx          15 Registered
>
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