By web based do you simply mean a UA that is *deployed* using the web
(http) or do you also mean that to include tunneling of media over 80/443?
Any Java based softphone could easily be turned into an applet, thus
satisfying the web-based part of your query.
An Active X component is nothing more
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] An example
= local
exten = 431,1,Dial,SIP/sipphone
Regovich, Timothy wrote:
Jason,
Include your sip and extensions files so people can take a look.
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Sent: Monday, February 23, 2004 10:25 AM
To: [EMAIL
Really?
Did you try
disallow=all
Allow=speex
Allow=gsm
Allow=alaw
?
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, February 23, 2004 2:21 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Codec Order / Preference
Two coffee cans and a tight string?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jamin W. Collins
Sent: Wednesday, February 18, 2004 12:06 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Room Monitor
On Tue, Feb 17, 2004 at 10:04:02PM -0800,
Turn sip debug on and forward the logs.
A 481 means that a dialog was not correctly established.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ariel Batista
Sent: Thursday, February 12, 2004 6:28 PM
To: Asterisk User List
Subject: [Asterisk-Users] Sip
Not ACK'ing an invite can be problematic for the statemachine. Without the
ACK, the Dialog is not in acorrect state.
As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same. There is no restriction that they must match. You can
change your offer in the
it.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maciek Kaminski
Sent: Wednesday, February 11, 2004 12:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Can't connect KPhone to asterisk
Regovich, Timothy wrote:
Not ACK'ing an invite can
Why does the FXO gateway treat a lack of RTP packets as a dropped call (and
what heuristic does it use to determine?)
Until the SIP UA sends an actual BYE message, the Dialog should still be
considered active, regardless of the RTP that may or may not be happening.
-Original Message-
with their support as a company. I
have still failed to get DTMF
bridging via RFC2833 working 100%. If anyone has had success with
Audiocodes FXO SIP gateways
and Asterisk, I would like to know the magic formula that makes all this
work. :)
Regovich, Timothy wrote:
Why does the FXO gateway treat
Jeremy,
There is one small flaw in your reasoning with the need to register. You
said :
You only need to register to Asterisk if you have a dynamic IP address
or you need to blow thru a firewall/NAT device
But this is not true if you want to maintain true presence information.
If you do not
Try RFC 3261
http://www.faqs.org/rfcs/rfc3261.html
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, February 04, 2004 12:45 PM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Sip flow diagram?
Does anyone have a
Or you could modify the logger and have all SIP messages set at a different
log level and have them go to a file (/var/log/messages/sip) for example.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geert Nijpels
Sent: Tuesday, February 03, 2004 11:38
Hello All:
Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
Is there something else that I need to be doing other than set the v flag
on my extension
]
Subject: Re: [Asterisk-Users] MeetMe Video option
Regovich, Timothy wrote:
Hello All:
Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
What video phone
: [Asterisk-Users] MeetMe Video option
Citeren Regovich, Timothy [EMAIL PROTECTED]:
Has anyone configured a meetme conference to use video?
I have successfully used video phones to talk through *, but I cannot seem
to get video when those phones dial into a meetme conference.
Cool, what devices
Asterisk decode all of the
video frames and create a 2x2 or 2x3 or 3x3 etc. mosaic, re-encode them
and send them to each client. That REALLY sounds like a pain to me, but
again, maybe it's doable.
Right now I'd be pretty happy with 2a though.
- Matt
Message: 3
From: Regovich, Timothy [EMAIL
Hi all,
Any advice on how to place a call from a SIP UA routed through *?
Do I just place a sip call to [EMAIL PROTECTED]:5060 ?
I am a little confused, since all of my Uas require registration for
presence information.
Thanks in advance,
Tim
Did anyone try compiling with optimizations off?
I seemed to noticed that the default flag was an O9 or something.
Try with -O1 or with -g ans see if it makes any difference.
Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dustin Knuttgen
Sent:
Hi all,
Does anyone have any experience with running a X100P card with * in an SMP
machine?
I have plugged the card into a 4way 2.4 GHz server, and the hardware config
seems ok -- the passthrough phone line works, the card has it's own IRQ on
CPU0, and /proc/zaptel/1 doesn't show any errors.
*
20 matches
Mail list logo
