[Asterisk-Users] Asterisk With Broadvoice

[James Taylor]

I have two Broadvoice "lines" and there's three people in the office.
Any way to:

1) "Pool" the connections for "trunking", where any one can get a "free"  
line?
2) Prevent more than 1 simultaneous call per "line"? (So I will not get  
hit for 3.9 cents a minute.

I'd like to use the country code AGI.

James

On Fri, 25 Feb 2005 15:52:28 -0500, Christopher McBee <cmcbee at rtctel.com>  
wrote:

> 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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>
>
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-- 
James Taylor
3505 Summerhll Road
Suite 11
Texarkana, Texas  75503
903-793-1953