Can HTB minimize latency better than CBQ?
Just curious,
Rich
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senas Jordanovic
Sent: Friday, July 23, 2004 1:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Priorizing of packets
This
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw
- Original Message -
From: Dr. Rich Murphey [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 1:28 PM
Subject: RE: [Asterisk-Users] Priorizing of packets
Can HTB minimize latency better than CBQ?
Just
Has anyone approached VoicePulse Connect about forwarding Caller Name?
Given the interest, it could be an effective way to differentiate service
among competitors.
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Bill Merriam
Sent: Friday,
whats been happening with me.
- Original Message -
From: Dr. Rich Murphey [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 8:01 PM
Subject: RE: [Asterisk-Users] Reverse number lookup
Has anyone approached VoicePulse Connect about forwarding
Caller Name
Differences in how poll() works is probably responsible.
Try this and see if it helps.
Cheers,
Rich
-Original Message-
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 16:03 -0500, Dr. Rich Murphey wrote:
That sounds like a bug. One should be able
Correct, I don't see that problem and it is loaded (by default).
In fact, I don't recall seeing it on 4.9, but I can't imagine that there
would be any significant difference in poll() for 4.10.
Rich
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From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
On Sun, 11 Jul 2004 at 15:39 -0500, Dr. Rich Murphey wrote:
You might check login class in login.conf for the user that invokes
asterisk. Setting cputime=unlimited may help
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arjan
Hi Brian,
On Sun, 11 Jul 2004 at 12:38 -0500, Brian K. West wrote:
start it with asterisk -vvvgc
Been there, done that. Nothing, no coredump. A lot of verbose
output while starting
At Astricon, I plan to cover QoS on FreeBSD using the pf
firewall's class based queuing. This includes implementing
classes to prioritize each of RTP, IAX, SIP, FTP, and others.
Within each class packets can be prioritized based on whether
TOS is set.
I'm wondering whether this should be a
How can one specify that codec selection should avoid transcoding if
possible?
The reason I ask is that when enum lookup succeeds and the destination only
accepts ULAW, the various transcodings seems to garble the audio if GSM,
ILBC, etc. are allowed.
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
George Pajari
Sent: Saturday, July 10, 2004 4:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Avoiding transcoding
Dr. Murphey:
How can one specify that codec selection should
I wish I had access to an OS X system. I could maintain more of the common
*BSD support if so.
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Stephan Wik
Sent: Monday, July 05, 2004 8:34 AM
To: [EMAIL PROTECTED]
Subject: Re:
Try changing definitions such as
static ast_mutex_t mylock;
to
AST_MUTEX_DEFINE_STATIC(mylock);
Look at asterisk/include/asterisk/lock.h for more info on lock
initialization.
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
How do you balance the number of active connections per server?
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nicholas Bachmann
Sent: Sunday, June 27, 2004 9:41 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
I have some patches that may help you. I'll isolate them and put them in
the asterisk bug database and post the location.
Cheers,
Rich
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of AK
Sent: Thursday, June 17, 2004 8:28 AM
To:
Asterisk on FreeBSD News
CURRENTLY:
Asterisk and libpri in Asterisk-current (CVS
head) build and run on FreeBSD 5.2.1.
The last major change (support for multiple
CPUs) has been incorporated into
Asterisk-current. Asterisk should now be
thread safe on FreeBSD and testing on dual
CPUs has just
Cd ../asterisk
patch your_diff_filename.diff
Cheers,
Rich
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Ed Devine
Sent: Tuesday, June 08, 2004 12:28
PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] HOW-TO
DIFF
How do I patch an * file if all I
Doesn't VOMIT capture calls?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Chris Albertson
Sent: Monday, June 07, 2004 5:29 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Network Sniffing Calls for recording
Many
If it fails to register, check the sip debug output for:
REGISTER sip:216.229.127.40 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060
If you see 0.0.0.0 in the 'Via' line, try using
nat=yes
externip=your external address
in your *global* section at the head of sip.conf.
I've searched but haven't
The status of the FreeBSD port of Asterisk
Changes this week:
Rich Nesse fixed compilation of app_intercom
and chan_oss. A solution to bug 1411,
recursive lock inititalization, was submitted
that seems to be stable on FreeBSD 5.2.1 in
terms of up-time. Libpri builds on FreeBSD
as well now.
Congradulations to the Asterisk gang on getting slashdotted!
http://slashdot.org/article.pl?sid=04/05/22/1840220
Cheers,
Rich
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-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: None
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk Cygwin Port.
Hello Everyone,
For the past few days I have been working on porting asterisk to
try asking on one of the cygwin lists. I may eventually run
across it, and I'll send you anything I can find.
Cheeers,
Rich
- Original Message -
From: Dr. Rich Murphey [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 6:15 PM
Subject: RE: [Asterisk-Users] Asterisk
: [Asterisk-Users] Asterisk Cygwin Port.
Hi,
Unfortunately yes. If these had worked it would have been much easier.
- Joshua Colp.
- Original Message -
From: Dr. Rich Murphey [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, May 21, 2004 6:15 PM
Subject: RE: [Asterisk-Users
It's in /usr/ports/misc/zaptel. For everyone else, there were some updates
to the driver recently by Maxim Sobolev (the maintainer) that can be
obtained by cvsup (see /usr/share/examples/cvsup/ports-supfile).
It works great with a single x100p card here as well!
Cheers,
Rich
-Original
If you take into consideration the need to pass DTMF to/from various SIP
providers, how do they rank?
I'd given up on g729 completely because none of the dtmfmode options seemed
to work with g729 for several sip service providers. GSM seems to work fine
in general.
Cheers,
Rich
-Original
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