And so the file said to the Brian... Let there be enlightenment:
strip(5);
That'll strip off the first 5... Characters... From the URI
Joshua Colp
On 11/10/05 1:25 PM, Brian C. Fertig [EMAIL PROTECTED] wrote:
Anyone with SER knowledge could you point me in a direction to setup SER to
This has already been discussed, you need
to upgrade your GCC to 3 or higher.
Joshua Colp
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rafael R. GV
Sent: Friday, November 04, 2005
11:47 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject:
Hello Harry,
This is rather the wrong list to ask this... since this is Asterisk, not
OpenPBX.org
Chan_exosip2 though is something I'm basically designing to have 3 operating
modes.
Full server: Most closely resembles chan_sip in that it acts as a B2BUA
Partial proxy: Extensions are mapped to
Hello Folks,
I thought Id make a sorta announcement as Ill be
in Montreal on
a partial vacation/partial hangout/partial meet and greet thing. I thought it
might be nice for all the people in the area, and perhaps those attending the
Meet Asterisk thing to get together for supper and
Hi Joseph,
Here's a basic entry for you that you should be able to adapt.
[mypeer]
Type=peer
Host=ip or hostname
Context=where to send the call
Disallow=all
Allow=ulaw
Insecure=very
The insecure=very causes Asterisk to not do any authentication and trust it
based on the IP.
Joshua Colp
On
Hello Ronald,
A 180 Ringing is something that should not have SDP because it's out of band
signaling of the exact status of the call, ringing. The PSTN Gateway should
return a 183 Session Progress if it wants to deliver inband audio progress.
Their SIP implementation doesn't look the best
Hi Scott,
To do what you want to do you do indeed need to use a peer entry, with the
IP address where INVITEs will come from specified as the host, and
insecure=very. Your OPTIONS though is being caused by qualify being turned
on somewhere.
Joshua Colp
-Original Message-
From: [EMAIL
You have to tell it the extension you want to pick up, it's not psychic.
Doing what you're doing now would give the application no extension.
Exten = _*99.,1,Pickup(${EXTEN:3}) should work, with usage being
*99extension to pickup
Joshua Colp
On 9/24/05 7:28 PM, Rich Adamson [EMAIL PROTECTED]
Hello,
Asterisk does not act as a SIP Proxy as you may have in mind. Each call is
treated independently, that is - codec capabilities of one call don't go to
the other one during a reinvite. Only the IP address and Port go.
Joshua Colp
-Original Message-
From: [EMAIL PROTECTED]
Hello Everyone,
For regular call pickup you can't really specify a pickup group number...
that's why it's set in the configuration.
For directed call pickup you need to have the latest CVS head as it uses an
API call that Kevin put in espically for me to use lastnight.
Joshua Colp
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