Re: [Asterisk-Users] i like my colors, thanks..

2005-04-21 Thread Jeffrey C. Ollie 
On Thu, 2005-04-21 at 08:39 -0500, Matthew Boehm wrote:
> Using most recent CVS-HEAD and my terminal keeps changing colors.
> 
> I'm using vt100 terminal emulation. How can I turn off asterisk's colors? Or
> at least turn off the black background. My normal terminal is white
> background, black font. But for some reason, asterisk is changing it to
> white font, black background.

Add '-n' to your command line. 'asterisk -h' will print out a list of
all of the command line switches that it supports.

Jeff


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Re: [Asterisk-Users] Cisco 7960/7960G

2005-04-19 Thread Jeffrey C. Ollie 
On Wed, 2005-04-20 at 13:48 +1130, Craig wrote:
> Can anybody tell me please what is the difference between a Cisco 7960
> and a 7960g phone.
> 
> I have been unable to determine the difference.

The "G" stands for global, which means that pictographs are used on the
right hand set of buttons (not the number pad) rather than English
words.  An original "G" from the factory includes a little sticker with
English words that fits over the buttons.

Jeff



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Re: [Asterisk-Users] Pretty Voicemail Docs

2005-04-13 Thread Jeffrey C. Ollie 
On Wed, 2005-04-13 at 10:57 -0500, Eric Wieling wrote:
> Has anyone written up "pretty" voicemail user docs?  I think voicemail 
> is so easy even my cat can use it.  However, my users are complaining 
> about lack of docs for voicemail.

I started some docs, haven't gotten around to finishing them.  You can
see what I came up with at:




Part of the problem with creating end-user documentation for Asterisk is
that it's just *too darn flexible*.  If you look though my docs, you'll
see that I used a lot of verbiage like "if your administrator did
this..." and "if your administrator did that...".

Plus, while the voicemail system probably had a clean and elegant
design, it has amalgamated a lot of patches and bug fixes that have left
the whole system quite a mess.  IMHO, the whole voicemail system needs a
ground-up redesign.

Jeff



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Re: [Asterisk-Users] Compile/modprobe issue

2005-04-12 Thread Jeffrey C. Ollie 
On Tue, 2005-04-12 at 20:29 -0700, Steven P. Donegan wrote:
> I'm attempting to put asterisk on a Soekris Net4801 with CRUX linux 
> (2.6.10 kernel patched as suggested). I get compile warnings and 
> modprobe failure on zaptel stuff:
> 
> zaptel: Unknown symbol crc_ccitt_table
> 
> I'm assuming that something needs to be in the kernel space that isn't - 
> any pointers to resolving this would be appreciated.

You need to have:

CONFIG_CRC_CCITT=m

set in your kernel config.



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Re: [Asterisk-Users] OT: mixing monitor files to stereo wav

2005-01-21 Thread Jeffrey C. Ollie 
With the help of some people on IRC I've gotten further with using
GStreamer:

gst-launch interleave name=int ! audio/x-raw-float,buffer-frames=256 !
audioconvert ! wavenc ! filesink location=x.wav { filesrc location=vm-
youhave.gsm ! audio/x-gsm,rate=8000 ! gsmdec ! audioconvert ! buffer-
frames-convert ! queue ! int.sink1 } { filesrc location=vm-
whichbox.gsm ! audio/x-gsm,rate=8000 ! gsmdec !  audioconvert ! buffer-
frames-convert ! queue ! int.sink2 }


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RE: [Asterisk-Users] OT: Headset for the Cisco 7960

2005-01-21 Thread Jeffrey C. Ollie 
On Fri, 2005-01-21 at 10:42 -0500, Nabeel Jafferali wrote:
> do you know how I could make that adapter if I
> wanted to use a single 2.5mm connector headset (like the kind used with
> cellphones and cordless phones)? Any idea what the pinout for that would
> be?
> 

http://www.mml.uni-hannover.de/einhorn/headset/index_e.html

Jeff



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Re: [Asterisk-Users] QOS / Cisco / Asterisk

2005-01-19 Thread Jeffrey C. Ollie 
Sorry for the late, late reply, but I don't follow the -users list
closely.

On Tue, 2005-01-04 at 10:43 -0600, [EMAIL PROTECTED] wrote:
>
> What's wrong with doing it by port? If it is possible that something
> else out there may use the same TOS flags as Asterisk, by prioritizing
> port 4569 (IAX2 protocol) you know for sure that the only packets in
> that queue are VoIP traffic. Also, what about your incoming traffic?
> Are the TOS flags correct there? I'm not saying that TOS is bad, just
> that as you've seen, it can get changed along the way. I'm using port
> number to separate traffic and it is working great. 

Well, in a sense, we are both correct.  You are looking at the problem
from the perspective of an edge router.  At the edge of your network,
you can't trust the incoming QOS markings, so you need to use an ACL of
some sort to differentiate priority traffic from non-priority traffic.

However, inside the network, when you can (mostly) trust that packets
have been generated with the correct QOS markings by the orginating
device, internal routers/switches can use the QOS marking (be it the
TOS, DiffServ markings, 802.1p priorities, etc.) to prioritize traffic.

I'd be willing to bet that switches (and maybe even some routers) can
prioritize based upon QOS markings more efficiently that they can run
packets through ACLs.  This is especially needed where traffic volumes
are large.

So, inside your network you need to examine the configuration of pretty
much every device to make sure that they don't mess with the QOS
markings where they aren't supposed to.

Jeff




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Re: [Asterisk-Users] QOS / Cisco / Asterisk

2005-01-04 Thread Jeffrey C. Ollie 
On Mon, 2005-01-03 at 13:53 -0600, Matt Schulte wrote:
> We're trying to PQ (Priority Queue) packets on a Cisco using ACL's. What
> we're trying to avoid is hardcoding the IP address in the ACL. We were
> trying to match by TOS set by Asterisk however it seems we've run into a
> snag where the packet TOS tends to get reset somewhere on our network.
> Has anyone had this issue? We're running Cisco everywhere inbetween
> (even the switches). Is there an alternative way to match these? We've
> thought of by port but that's kind of ad-hoc IMHO.

If the TOS is getting reset somewhere out there you need to go through
all of your switches and make sure that none of them are messing with
the TOS.  Unfortunately doing QOS on Cisco switches is a black art as
the necessary commands depend on the hardware and the IOS version (or
CatOS version if you are unlucky).  Check the documentation for your
switches for the "mls qos trust" command.

Cisco routers, on the other hand, don't mess with IP TOS/DSCP labels
unless you specifically ask them to.

Jeff



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Re: [Asterisk-Users] Cisco 7960 SIP + 7914

2004-12-15 Thread Jeffrey C. Ollie 
On Wed, 2004-12-15 at 11:54 -0600, Matt Schulte wrote:
> I found a few mentions of the 7914 being used with Asterisk, these all
> covered SCCP/skinny though. Does anyone know if the 7914 can even be
> used with SIP? If so, any pointers? Is it a services thing? Anyone get
> the operator (line/extension status) to work with it. Thanks for the
> help, Cisco doesn't even mention ANYTHING about SIP + the 7914.

The 7914 is not supported by Cisco's SIP code. If you look at the data
sheet under "System Requirements" is says that you need Cisco
CallManager, which implies SCCP/skinny:

http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html

Jeff


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[Asterisk-Users] Voicemail User Reference Guide

2004-12-10 Thread Jeffrey C. Ollie 
I'm developing a user reference guide for Asterisk's voice mail
application.  Note, that this is going to be a _user_ guide, not an
administrator guide.  The current document is rather rough, but it
should give you an idea of the direction that I'm headed.  I'd
appreciate constructive criticism.

The document can be found at:

http://www.ocjtech.us/vm.pdf

The DocBook XML source files can be found at:

http://www.ocjtech.us/vm.xml
http://www.ocjtech.us/fdl.xml

Jeff Ollie



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Re: [Asterisk-Users] Cisco Asterisk Integration

2004-11-30 Thread Jeffrey C. Ollie 
On Wed, 2004-12-01 at 11:43 +0800, Dinesh wrote:
>
> 046 Looking for 3000 in from-sip-external
> 047 Reliably Transmitting (no NAT):
> 048 SIP/2.0 404 Not Found

Looks to me like extension 3000 is not in the "from-sip-external"
context.  Check your dialplan.

Jeff



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Re: [Asterisk-Users] OT: mixing monitor files to stereo wav

2004-11-28 Thread Jeffrey C. Ollie 
On Sun, 2004-11-28 at 23:01 +, Stefan Reuter wrote:
> 
> i am looking for a tool to merge the two wav files of a monitored call
> into one. soxmix does that well but actually merges the two channels.
> I would prefer a solution that creates a stereo wav file of the two mono
> files so you have the called party on one (e.g. left) channel and the
> calling party on the other (e.g. right).

This doesn't really help right now, but it looks like GStreamer will
soon be able to help you with a command like this:

gst-launch interleave name=int ! wavenc ! filesink location=/tmp/x.wav
{ filesrc location=ch2.gsm ! gsmdec ! queue ! int.sink1 } { filesrc
location=ch2.gsm ! gsmdec ! queue min-threshhold-buffers=1 ! int.sink2 }

Unfortunately the interleave plugin isn't currently fully implemented,
so this command bombs out with an error.  No one was awake on the IRC
channel to help out so hopefully they'll release a new version soon that
has the necessary bits implemented.

Jeff



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Re: [Asterisk-Users] [PATCH] DUNDi for 1.0.2

2004-10-27 Thread Jeffrey C. Ollie 
On Wed, 2004-10-27 at 23:31 -0400, William Suffill wrote:
> Great job Jeff. Lets hope the dbscret can be patched up soon too but
> this is a great leap forward.

There are new versions of the patches on Mantis that appear to fix the
dbsecret problems.

Jeff


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[Asterisk-Users] [PATCH] DUNDi for 1.0.2

2004-10-27 Thread Jeffrey C. Ollie 
I have backported pbx_dundi.c (rev 1.12 from CVS HEAD) to the 1.0.2
release.  The patch has been posted to Mantis at
.

Jeff Ollie


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RE: [Asterisk-Users] IP Phone that OFFICIALLY support Asterisk

2004-10-20 Thread Jeffrey C. Ollie 
On Wed, 2004-10-20 at 17:27 -0400, Jim Van Meggelen wrote:
>
> And as for Call Manager? I predict that they will be officially
> Asterisk-compliant in . . . hmmm . . . I'll say roughly five years or
> so. Possibly far sooner if they yank their heads out of their asses and
> grab a clue. Asterisk is destined to do for telecom what Linux did for
> the OS. In five years, many big names will be waving the Asterisk flag
> in the exact same way they are with Linux. Mark my words.

Actually, CallManager 4.0 supports SIP trunks which interoperates with
Asterisk quite well.

And I've heard rumors that Cisco is porting CallManager to Linux, FWIW.

Jeff Ollie



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RE: [Asterisk-Users] IAXy setup

2004-10-17 Thread Jeffrey C. Ollie 
On Sun, 2004-10-17 at 00:20 -0400, Jim Van Meggelen wrote:
>
> Nevertheless,
> it's kinda not proper to deliver an ethernet device that is not labeled
> with it's MAC address. Why should we have to go through any kind of
> trouble to determine this? I say it should be on the unit.

The MAC address label should have the MAC address encoded as a bar code
as well. When you're deploying hundreds of devices that and need to
program their MAC addresses into your * box (or whatever) it helps to
have the bar codes and a bar code reader.

Jeff Ollie



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