Hi All,
My version of asterisk is Asterisk CVS-HEAD-03/07/05-17:14:42
I get this error with broadvoice.
-- Executing Dial("SIP/10.217.84.12-0816c7d0",
"SIP/broadvoice/011612464823xx") in new stack
Mar 7 18:52:44 NOTICE[794]: app_dial.c:936 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("SIP/10.217.84.12-0816c7d0", "") in new stack
== Spawn extension (default, 2011612464823xx, 2) exited non-zero on
'SIP/10.217.84.12-0816c7d0'
owl*CLI> exit
does anyone know what I am doing wrong?
thanks
Dinesh.
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wolfgang S.
Rupprecht
Sent: Monday, March 07, 2005 2:26 PM
To: [email protected]
Subject: Re: [Asterisk-Users] BroadVoice configuration changes for Outbound
[EMAIL PROTECTED] (Dan Weber) writes:
> On Sat, 5 Mar 2005, Wolfgang S. Rupprecht wrote:
>> Does broadvoice participate in e164.{arpa,org,info}?
>>
> Yes
>> Does this change mean that non-customers can't call broadvoice
>> customers with a pure SIP call by routing the call to
>> sip.broadvoice.com?
>>
> Calls can be made to broadvoice phones by <phonenumber>@sip.broadvoice.com
>> (From a security standpoint what is the difference between calling the
>> BV customer directly vs over the TELCO lines? Perhaps I'm missing
>> something, but better/cheaper/faster to cut out the telco middleman.)
>>
> Much cheaper over internet vs. telco.
That's great news! I had a sinking feeling when I heard the words
"authenticated invite".
Unfortunately some large voip companies (cough cisco) are locking down
their sip servers to only talk to established peers. Perhaps I'm
missing something crucial, but these companies still have DID numbers
for their employees, so locking down the sip server just forces the
call to go out via the PSTN.
So are BV customers listed in the in e164.org dns zone (or some other
publicly accessible routing database)? I would love to have some way
to bypass the telco when calling friends without having to put a
by-hand entry into asterisk for each person that can accept direct
calls via some voip proxy.
-wolfgang
--
Wolfgang S. Rupprecht http://www.wsrcc.com/wolfgang/
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