[Asterisk-Users] Asterisk With Broadvoice

asterisk at billwho.com asterisk at billwho.com
Fri Feb 25 08:20:35 MST 2005


Great!  It works now!!  Thanks so much.

Roger Hanson wrote:

>
> ----- 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
>
>
>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users





More information about the asterisk-users mailing list