[Asterisk-Users] Asterisk With Broadvoice
Roger Hanson
roger at makarios.us
Fri Feb 25 14:11:00 MST 2005
----- Original Message -----
From: "Robert Webb" <asterisk at ropeguru.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>; <asterisk at billwho.com>
Sent: Friday, February 25, 2005 2:49 PM
Subject: Re: [Asterisk-Users] Asterisk With Broadvoice
On Fri, 25 Feb 2005 14:42:09 +0000
"asterisk at billwho.com" <asterisk at billwho.com> wrote:
> 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
>>
Mine ONLY works both directions when I use a normal registration string.
And remember, don't use the password you signed up with on their
website. They email you a different password you need to use in your
Asterisk configurations.
I know some people have to use the funky registration string, but it
wouldn't work for me (and some others). Also, I know of some others
that couldn't get it to work without the line: insecure=very
Here's my sip:
register=myphonenumber:mypassword at sip.broadvoice.com
[myphonenumber]
type=friend
secret=mypassword
regexten=myphonenumber
insecure=very
host=sip.broadvoice.com
fromuser=myphonenumber
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=from-pstn
canreinvite=yes
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