date:20031223

On Tue, 2003-12-23 at 10:52, tony banks wrote:
 Hello 
 
 When I tried loading TDM400P module using insmod command, I get following error 
 messages. Is there some problem with my asterisk installation. Please advise. Thanks 
 Tony
 
 $insmod wcfxs
 Using /lib/modules/2.4.20-8/misc/wcfxs.o
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_ec_chunk
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_unregister
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_hooksig
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_transmit
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_receive
 /lib/modules/2.4.20-8/misc/wcfxs.o: unresolved symbol zt_register
 
 

That looks like a Red Hat kernel that has a local root exploit iirc so
you may wan to upgrade that one. If you haven't done that already, make
sure you have the kernel sources installed. Get a fresh copy of zaptel,
libpri  asterisk from cvs and then try again. I think the error means
that you are trying to load zaptel modules that were build for a
different kernel.

Patrick

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[Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread Adthrawn

Hi,

I'm a newbie to the list, but have been screwing around with Asterisk 
for the last 6 months or so (on a purely experimental basis so far). 
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm 
unsure where the line is drawn in terms of Linux issues or Asterisk 
issues.

At present, I have to manually start Asterisk from the command line, 
but I'd like to have it automatically start up (and in the correct 
mode) at startup.

For now, the server is running as a workstation, so I only need it to 
run as a background daemon, but in the near future, we're going to run 
Asterisk of a dedicated racked server, which we would only want to run 
Asterisk, and there bare minimums required - as far as I'm aware, you 
could start Asterisk very early on in the boot-up process.

Can anybody guide me in configuring the system to start Asterisk from 
bootup... Probably a highly remedial question - but you've got to start 
somewhere!

Regards,
Ad.
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[Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

2003-12-23 Thread Adthrawn

Hi,

Has anybody been successful in running the 7914 expansion unit for the 
Cisco 7960G IP phone? For anybody unaware of what the expansion unit 
does, it provides 14 additional buttons, with an LCD display. The idea, 
is that with an expansion unit (a 7960 can take upto 2 of these units), 
a user can either assign more speed-dial's, or can monitor line 
status/account status. So, you can either register a speed-dial or 
register another account.

The problem I've found so far, is that speed-dials are not programmed 
on the phone, but are instead handled by the Call Manager software (not 
on a user basis, but on a phone, MAC address basis). Likewise, plugging 
the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red 
lights (the buttons also light-up red, blue or green), which according 
to the hidden technical documentation, indicates that the Call Manager 
is not registering the unit. I can't work out if it's short of firmware 
embedded in the Call Manager, whether it's searching for a 
configuration file on the TFTP (Cisco phones need a TFTP to get their 
settings and SIP firmware), whether it's not happy with the phone being 
a SIP version, or whether I'm doing something wrong.

I've had to learn about the 7960's configuration the hard way, and 
despite their useless technical documents, have managed to configure 
most settings.

There's quite a bit of extra configuration for the 7960 I'd love to get 
to, and would like help or advice on. Things like directory services, 
screen logo, the 7914 and more!

If anybody is interested, I have resources and files to; convert from 
Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail 
indicator lamp, special key combinations to reset the unit (without 
pulling the plug out) and locking/unlocking the preferences, 
configuring the voicemail speed-dial

Any help or advice, please let me know!

Regards,
Ad.
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RE: [Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread David J Carter

Hi,

In rc.local I added the line /etc/rc.d/run-asterisk


I then created a small script of 2 lines called run-asterisk

#!/bin/sh
/usr/sbin/asterisk

do a chmod 755 on the file and reboot.

The Asterisk server then starts at every reboot.


Regards


Dave


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adthrawn
Sent: 23 December 2003 12:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto Starting Asterisk

Hi,

I'm a newbie to the list, but have been screwing around with Asterisk
for the last 6 months or so (on a purely experimental basis so far).
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm
unsure where the line is drawn in terms of Linux issues or Asterisk
issues.

At present, I have to manually start Asterisk from the command line,
but I'd like to have it automatically start up (and in the correct
mode) at startup.

For now, the server is running as a workstation, so I only need it to
run as a background daemon, but in the near future, we're going to run
Asterisk of a dedicated racked server, which we would only want to run
Asterisk, and there bare minimums required - as far as I'm aware, you
could start Asterisk very early on in the boot-up process.

Can anybody guide me in configuring the system to start Asterisk from
bootup... Probably a highly remedial question - but you've got to start
somewhere!

Regards,
Ad.

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RE: [Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread mikeu

I use http://cr.yp.to/daemontools.html.  Besides starting asterisk on boot
up it keeps an eye on the process and restarts asterisk if it crashes.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J Carter
Sent: Tuesday, December 23, 2003 6:38 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Starting Asterisk

Hi,

In rc.local I added the line /etc/rc.d/run-asterisk

I then created a small script of 2 lines called run-asterisk

#!/bin/sh
/usr/sbin/asterisk

do a chmod 755 on the file and reboot.

The Asterisk server then starts at every reboot.

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adthrawn
Sent: 23 December 2003 12:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto Starting Asterisk

Hi,

I'm a newbie to the list, but have been screwing around with Asterisk
for the last 6 months or so (on a purely experimental basis so far).
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm
unsure where the line is drawn in terms of Linux issues or Asterisk
issues.

At present, I have to manually start Asterisk from the command line,
but I'd like to have it automatically start up (and in the correct
mode) at startup.

For now, the server is running as a workstation, so I only need it to
run as a background daemon, but in the near future, we're going to run
Asterisk of a dedicated racked server, which we would only want to run
Asterisk, and there bare minimums required - as far as I'm aware, you
could start Asterisk very early on in the boot-up process.

Can anybody guide me in configuring the system to start Asterisk from
bootup... Probably a highly remedial question - but you've got to start
somewhere!

Regards,
Ad.

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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

 Packet 50 - Delay 50ms
 Packet 51 - Delay 5ms
 Packet 52 - Delay 5ms
 Packet 53 - Delay 50ms
 Packet 54 - Delay 5ms
 Packet 55 - Delay 5ms
 
 Is there anyway to space them out evenly at 20ms??
  
  
  The 20 ms is not the inter-packet timing, its the relative content of what's
  within the packet. In other words, the packet contains 20ms of encoded voice.
  
  If the inter-packet times (delays) are large, as they would seem to be
  in your example, then something else is not right. Possibly a half-duplex
  ethernet connection, something else running on the server, router buffers,
  etc.
  
  On a typical * -- C7960 local call, I generally see from 1ms to 20ms
  inter-packet delays. Seldom (if ever) anything above 20ms.
  
 
 I gather from your reply that there are recommendations regarding the 
 ethernet connection on your Asterisk server? half-duplex seems bad.
 Could you elaborate a bit on that?

Yes, half-duplex ethernet connections can cause significant problems
depending on the actual load. In very general terms, a half duplex
ethernet interface can run up to about 20% utilization before collisions
occur, whereas a full duplex connection can run close to 100% without
dropping packets. Those rough numbers apply to both 10 meg and 100 meg
ethernets.

If a collision or dropped packet occurs (in a voip udp environment) there
is no way to retransmit the missing/damaged packet. Missing one packet isn't
a big deal, but if you have collisions and/or dropped packets, there is a
very high probability that lots of packets will be dropped. If too many
are dropped, you'll hear the result in the undecoded voice as choppy 
voice.

For whatever reason, most unix systems (and MS systems for that matter)
do not give you a convenient way to configure (or even check) how your 
ethernet adapter negotiates the connection. There are no official
standards as to how the negotiation process determines half vs full,
and systems get it wrong about 50% of the time. (As professional network
performance consultants, we've diagnosed a very large number of problems
like this in corporations around the US over the last ten years. Think in 
terms of a unix system trying to negotiate half vs full at the exact same 
time as the switch is doing the same thing without actually communicating 
to the opposite end of the cable.)

If the ethernet traffic is low, no one actually notices the problem. But,
as traffic increases (multiple RTP sessions, etc) the problem begins to
occur and the average technical person doesn't have a clue what is really
going on. What makes it difficult to identify/diagnose is that each time
the system is rebooted (and each time a Cat 5 cable is disrupted), the
half vs full negotiation happens again and (as mentioned) 50% of the time
one end gets it wrong. Therefore the performance problem tends to come
and go, and support folks typically don't associate the performance
issue with the actual half/full problem. (In larger companies, the network
support person might reboot a switch without the * support person
knowing it, and suddenly the * support person has a problem for which
he can't identify what happened.)

Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
ethernet would handle roughly 20-25 rtp sessions before bumping into the
problem (your milage may vary). The majority of the folks on this list
seem to be running home/soho systems and would likely never run into the
issue. But the heavier users will.

What makes this half/full problem even more difficult to diagnose is that
many of these systems have other functions running on them (eg, up2date,
remote database calls, web activity, broadcasts) that can consume a fair
amount of ethernet bandwidth, and the support person is so highly focused
on asterisk they forget some other activity might be impacting their voip
quality. Invariably, a Cat 5 cable disruption or reboot (or something
else) happens at the same time the support person makes a programming or
parameter change, and the person jumps to the conclusion they fixed a
problem with their change when in fact the problem was with their ethernet
connection.

To ensure one never gets bit by the issue, simply ensure that all ethernet
interfaces between the asterisk system and the sip phones are statically
defined as full-duplex. (Good luck in finding the utilities that let you
do that on Linux systems. They are out there, but not easy to find.)

The sip phone's negotiation of half vs full is less of an issue as generally
the most traffic it sees is one RTP session. But, to obtain maximum smoke
and ensure highest quality, the phones should be locked at full duplex as
well.

Rich



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RE: [Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread Bisker, Scott (7805)

An even better way to get asterisk started is to use the init scripts provided with 
asterisk and the zaptel kernel modules.

cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk
cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel

Then do the proper linking, etc to get asterisk to start in your current run level.

-sb



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: Tuesday, December 23, 2003 7:38 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Starting Asterisk


Hi,

In rc.local I added the line /etc/rc.d/run-asterisk


I then created a small script of 2 lines called run-asterisk

#!/bin/sh
/usr/sbin/asterisk

do a chmod 755 on the file and reboot.

The Asterisk server then starts at every reboot.


Regards


Dave


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adthrawn
Sent: 23 December 2003 12:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto Starting Asterisk

Hi,

I'm a newbie to the list, but have been screwing around with Asterisk
for the last 6 months or so (on a purely experimental basis so far).
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm
unsure where the line is drawn in terms of Linux issues or Asterisk
issues.

At present, I have to manually start Asterisk from the command line,
but I'd like to have it automatically start up (and in the correct
mode) at startup.

For now, the server is running as a workstation, so I only need it to
run as a background daemon, but in the near future, we're going to run
Asterisk of a dedicated racked server, which we would only want to run
Asterisk, and there bare minimums required - as far as I'm aware, you
could start Asterisk very early on in the boot-up process.

Can anybody guide me in configuring the system to start Asterisk from
bootup... Probably a highly remedial question - but you've got to start
somewhere!

Regards,
Ad.

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Re: [Asterisk-Users] Callwaiting / limits?

2003-12-23 Thread Stephen J. Wilcox

   I'm using grandstream phones, when on a call and a second call comes in
 the
  call waiting indication is to play ringing which means you cant actually
 hear
  your original call. I want to stop this but cant, heres my options
 
  1. Change the callwaiting indication, I assume this is produced by the
 phone and
  in the case of grandstream there seems to be no way to control this.
 
  2. Use of incoming/outgoing limit in sip.conf. This works okay except
 there is
  no 'absolute limit' type option, meaning that if i place an outbound call
 from
  my grandstream it is possible to send a new incoming call in and we have
 the
  call waiting again.
 
  I assume others have found this, whats the solution?
 
  Steve
 
 
 Hi Steve,
 
 The incominglimit applies to both incoming and outgoing calls, so long as
 I'm on the phone, any incoming call gets sent to voicemail. Use the sip
 show inuse on the CLI to check the inuse counter is being incremented when
 on a call, whether receiving or outgoing.
 
 Is anybody else having this problem ?

Ok have looked a bit closer, it works for ordinary calls, my problem is actually 
with queuing. The call isnt going to the phone (at least the inuse counter stays 
at 1) so it must either be asterisk adding the ring sound to the stream (doesnt 
seem likely) or the queue app is ignoring incominglimit

i've just started to look at the code to see if i can spot whats going on but 
this is my first time in doing so for asterisk so i'm not familiar at all with 
its inner working! :)

Steve

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[Asterisk-Users] codes/grandstream/PRI.. few questions :)

2003-12-23 Thread vocalvoip

Hi Guys..

Just wondering if someone could help me with a few questions please. were currently 
using the ulaw codec with our grandstream/iconnect/asterisk setup and its working 
pretty good except for the fact it downloads heaps. Does anyone know a good site to 
get referances to how much each codec downloads/quality etc etc ? Ive tried using that 
g723 codec but i have have problems as soon as a i dial..

my next question.. :) does anyone know howto fix the grandstream 484 errors you get 
sometimes when you dial ? i had a look at they rekon to put early dial on.. which just 
makes things worse heh. 
They'd be a cool little phone except for this problem. 

Lastly were looking at getting a PRI or something to handle 30 lines.. I know digium 
sells hardware to do this, has anyone in australia gotten good results from doing this 
kind of setup ?? also what are the restrictions in regards to caller id and that sort 
of stuff in aus? do is all work ?


thanks heaps everyone :)

Merry Christmas

Justin
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RE: [Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread Thorsten Lockert

make config does both the copy and the neccecary linking...

Thorsten 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott
(7805)
Sent: Tuesday, December 23, 2003 8:50
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Starting Asterisk

An even better way to get asterisk started is to use the init scripts
provided with asterisk and the zaptel kernel modules.

cp /usr/src/asterisk/init.asterisk /etc/init.d/asterisk
cp /usr/src/zaptel/init.zaptel /etc/init.d/zaptel

Then do the proper linking, etc to get asterisk to start in your current run
level.

-sb

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: Tuesday, December 23, 2003 7:38 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Auto Starting Asterisk

Hi,

In rc.local I added the line /etc/rc.d/run-asterisk

I then created a small script of 2 lines called run-asterisk

#!/bin/sh
/usr/sbin/asterisk

do a chmod 755 on the file and reboot.

The Asterisk server then starts at every reboot.

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adthrawn
Sent: 23 December 2003 12:18
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Auto Starting Asterisk

Hi,

I'm a newbie to the list, but have been screwing around with Asterisk
for the last 6 months or so (on a purely experimental basis so far).
I'm not a linux expert by any stretch, (I'm a Mac OS X user), so I'm
unsure where the line is drawn in terms of Linux issues or Asterisk
issues.

At present, I have to manually start Asterisk from the command line,
but I'd like to have it automatically start up (and in the correct
mode) at startup.

For now, the server is running as a workstation, so I only need it to
run as a background daemon, but in the near future, we're going to run
Asterisk of a dedicated racked server, which we would only want to run
Asterisk, and there bare minimums required - as far as I'm aware, you
could start Asterisk very early on in the boot-up process.

Can anybody guide me in configuring the system to start Asterisk from
bootup... Probably a highly remedial question - but you've got to start
somewhere!

Regards,
Ad.

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Re: [Asterisk-Users] tor2 does not load

2003-12-23 Thread Steve Underwood

Eduardo Goncalves wrote:

On Mon, 22 Dec 2003 15:48:37 -0600
Steven Critchfield [EMAIL PROTECTED] wrote:
 

asterix:~# modprobe tor2
Zapata Telephony Interface Registered on major 196
Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000
irq 7 Did not get DONE signal. Short file maybe??
 

Just a guess, but maybe your module file is corrupted. Have you tried
recompiling the module? If that doesn't work, try the standard move
the card to a different slot. Sometimes cards can become belligerent
and will not wake up until they have been initialized in a different
slot. This is not a digium specific trick, but a problem I have had
with other cards. 
   

	I tried recompiling, but the error is the same. I tried in another
machine also. 

	It's strange that lscpi now shows a line that I've never seen before:

asterix:~# lspci
00:03.0 Bridge: PLX Technology, Inc.: Unknown device d00d (rev 01)
 

The dood in question is Jim Dixon, one of the developers of the tormenta 
2 card. :-) Thats is your Tormenta 2 card.

Regards
Steve




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RE: [Asterisk-Users] DID trunks -- equipment requirement

2003-12-23 Thread Tim Thompson

You should be able to just order Trunk Lines.

They are also known as ground start lines.  They are usually for
incoming only so you would have something like 4-5 Trunk lines for the
incoming DID's and the rest would be regular pots lines.

In your CAC, you would take the Trunk lines and they would come in on
the FXS channels and the POTS lines would come in on the FXO channels.

 
In our area, Trunk lines run about $29-$35 each and then you pay for the
DID's.

Hope it helps.


Tim Thompson
Commercial Sales Engineer
http://www.amatechtel.com
(806) 722-2227


-Original Message-
From: Don Pobanz [mailto:[EMAIL PROTECTED] 
Sent: Monday, December 22, 2003 4:42 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [Asterisk-Users] DID trunks -- equipment requirement

On Monday, December 22, 2003 3:40 PM, john lawler 
[SMTP:[EMAIL PROTECTED] wrote:
 Hi guys,

 I posted a somewhat similar question about a month ago and got a
 thoughtful resonse from Steven Critchfield, but I've got a quick
 follow
 up question to it.

 I'm looking to setup a 16 extension / 10-14 phone line Asterisk
 install
 for a customer who would like to have DID numbers for the extensions,

 since they're currently on Centrex and already have the 1-to-1
 correspondence.  Since I'm in a less populated area of the country,
 SBC
 doesn't seem to have much in the way of fractional T1 products (on 
the

 scale that we need them) available,

Have you asked for a full T1 but with just 10-14 DID/DOD trunks? We can 
not get fractional T1 here but on a full T1 we can add anywhere from 1 
- 24 trunks. So we pay one amount per month for the T1 and on top of 
that we pay another amount times the number of trunks we have.

I know this didn't exactly address your questions. For your primary 
question I believe that your would need different type of channels in a 
channel bank than FXOs. DPTs (Dial pulse) terminating come to mind, but 
that may be wrong.

Don Pobanz

so I think my only option for DID
 is
 to use (analog) DID trunks for incoming calls and POTS lines for
 outbound calls.

 I'm familiar w/ POTS lines and have already done limited testing w/ a
 CAC channel bank equipped with FXO cards and that works fine.  What
 I'm
 concerned about is the DID trunks.  I've been told they have no
 dialtone
 and of course you can't place calls on them, but can receive calls.

 My question is, in general, should my CAC channel bank w/ the FXO
 cards
 that work on POTS lines work okay w/ analog DID trunks from the phone

 company?  Might I have to purchase additional equipment to handle the

 DIDs (going into one of two Digium T1 cards I have in the Asterisk
 box)?  Would they be different cards to plug into the CAC channel
 bank?
 Something totally different?

 Sorry to bring what I know is a rather off-topic question here, but
 the
 SBC guys don't like to help with customer education so much.  As
 always,
 I appreciate all of your expertise and patience with me and the other

 new guys.

 John Lawler

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[Asterisk-Users] gnophone transfer

2003-12-23 Thread Anton Yurchenko

hello,

Is there a way to transfer the call via gnophone, without calling other 
user and pressing conf on both calls, it seems that all traffic is still 
going through the gnophone, not that optimal i guess.

thanks

--

Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
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[Asterisk-Users] Music On Hold in Conference room?

2003-12-23 Thread Michael Graves

Hello All,

Does anyone here know how I might provide music into a conference room
when there is only one participant. Dead silence tends to confuse
non-techies who think that they've done something wrong, even after the
entry announcement.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc.  [EMAIL PROTECTED]
 FWD 54245

Lawyers, guns and money can't get me out of this. - Warren Zevon
 
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704


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[Asterisk-Users] sendmail problems

2003-12-23 Thread jr.richardson

Hello,

I'm having some * and sendmail integration problems, probably because i don't know too 
much about sendmail.  My server crashes when I forward voicemail from one * voicemail 
box to another, everything else works.  E-mail notification works on all boxes when 
new mail arives, the problem only seems to occur during this forwarding function.  
It's a difficult problem to troubleshoot.  If I start * -gc, the server doesn't crash, 
just hangs up for about 60 seconds then completes the task, so i can't seem to get a 
core dump to dive into the specifics of what's going on.  I'm not sure how to debug 
sendmail to look at that side.  If someone would be kind enough to e-mail me some 
sample sendmail.cf files, I may be able to see if I'm not configure properly.  I've 
been reading the sendmail.org site but this application is really archain and 
difficult for me to understand enough to fix it myself.  Thanks in advance.

JR

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Re: [Asterisk-Users] Asterisk + CRM

2003-12-23 Thread Jonathan Tew

We're starting to integrate * with our customer service software.  
Basically we're pulling off events from the management interface.  We're 
also making some small patches to the code to deliver more events about 
the channel variables, etc. 

Anton Yurchenko wrote:

Hello,

Anyone aware of any CRM products projects that intagrete with *? Or 
that integrate with any telephony products? Is there some open API for 
such integration, or are they all proprietory?

Thanks



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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

2003-12-23 Thread Clif Jones

Interesting.  For the record, the MultiTech MVP-130 comes with a default 
setting
of 60ms packets on all of its supported codecs.  I changed the packet 
sizes to
20ms because I had never heard of anyone using such large sample sizes.

Andres wrote:

On Monday 22 December 2003 19:58, Rich Adamson wrote:
 

On Monday 22 December 2003 16:37, Andres wrote:
 

On Monday 22 December 2003 15:36, Rich Adamson wrote:
   

I have a question regarding the Asterisk Packet Time for SIP Calls.
It is hardcoded at 20ms but when I do an RTP Analysis on a stream
it is clear that these packets are not spaced out at 20ms.  In
general you see something like:
Packet 50 - Delay 50ms
Packet 51 - Delay 5ms
Packet 52 - Delay 5ms
Packet 53 - Delay 50ms
Packet 54 - Delay 5ms
Packet 55 - Delay 5ms
Is there anyway to space them out evenly at 20ms??
   

The 20 ms is not the inter-packet timing, its the relative content of
what's within the packet. In other words, the packet contains 20ms of
encoded voice.
If the inter-packet times (delays) are large, as they would seem to
be in your example, then something else is not right. Possibly a
half-duplex ethernet connection, something else running on the
server, router buffers, etc.
On a typical * -- C7960 local call, I generally see from 1ms to 20ms
inter-packet delays. Seldom (if ever) anything above 20ms.
 

Thanks for your Input Rich.  I went ahead and tested this on our
production servers and sure enough the inter-packet times are 20ms. 
There must be something happening with our LAB Asterisk.  It could be
the CBQ traffic shaping software we have running on it.  I will fiddle
around with it to see if it changes anything.

Thanks!
Andres
   

Ok...after some more testing, the traffic shaping software was not the
culprit.  It turns out that if the UA is configured for 60ms of voice,
then Asterisk will show this strange behaviour.  If we set the UA for
20ms, then all works well.
 

Cool!

How did it get set to 60ms?
   

The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the 
transmit packet size to 60ms (or multiple other values).  Asterisk will 
receive 60ms and transmit 20ms times 3 packets, andit works quite well.  In 
any case our SPA2000 problem was unrelated to the packet time.

Regards,
Andres 
 

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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

2003-12-23 Thread Joel Maslak

On Tue, 23 Dec 2003, Rich Adamson wrote:

 If a collision or dropped packet occurs (in a voip udp environment) there
 is no way to retransmit the missing/damaged packet. Missing one packet isn't
 a big deal, but if you have collisions and/or dropped packets, there is a
 very high probability that lots of packets will be dropped. If too many
 are dropped, you'll hear the result in the undecoded voice as choppy
 voice.

Actually, collisions occur at Layer 2, not Layer 3, and the layer 2
hardware automatically resends packets involved in a collision - layer 3
is never aware of it happening (although it may cause additional delay).
Eventually the ethernet card will give up if too many collisions occur
during retries, but this is very rare in practice unless the network is
*VERY* loaded.

 Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
 ethernet would handle roughly 20-25 rtp sessions before bumping into the
 problem (your milage may vary). The majority of the folks on this list
 seem to be running home/soho systems and would likely never run into the
 issue. But the heavier users will.

For a duplex mismatch, my experience is that if one end on a 100 Mb/sec
link is half and the other is full, bandwidth is limited to about 8 Mb/sec
max.  This is based on some tests I've accidentally conducted.  If you try
to send 9 Mb/sec over that link, yes, some packets will get dropped as
they simply won't fit.  (But I do agree that for a half-half link, you can
get about 20 Mb/sec)

-- 
Joel
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Re: [Asterisk-Users] Asterisk + CRM

2003-12-23 Thread CW_ASN - Gus

Which events do you refer?

Regards,

Gus

- Original Message - 
From: Jonathan Tew [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 12:25 PM
Subject: Re: [Asterisk-Users] Asterisk + CRM

 We're starting to integrate * with our customer service software.  
 Basically we're pulling off events from the management interface.  We're 
 also making some small patches to the code to deliver more events about 
 the channel variables, etc. 

 Anton Yurchenko wrote:

  Hello,

  Anyone aware of any CRM products projects that intagrete with *? Or 
  that integrate with any telephony products? Is there some open API for 
  such integration, or are they all proprietory?

  Thanks

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[Asterisk-Users] PBX Functionality How-to

2003-12-23 Thread Christopher J. Wolff

Hello,

I had a partner of mine present a Centrex 21 brochure and ask how many of
those features can I fulfill.  There is nothing out of the ordinary, it's
stuff like call hold, call forward, 3-way calling, etc.  Has anyone
assembled a how-to that shows how to configure PBX or Centrex type
functionality?  I found one in the voip-info wiki but only a couple of
topics were filled out.

Regards,
Christopher J. Wolff, VP CIO
Broadband Laboratories, Inc.
http://www.bblabs.com



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[Asterisk-Users] Capi Dial outgoing msn?

Hi all,

I am trying to get Capi Dial to use a specific outgoing msn. I can't get
it to work. If I make a test call to 0703241494 (same isdn line, just
one of the other numbers) I don't get CLID at all. Any ideas?

; use 0703241432 as outgoing msn
exten = _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r)

in capi.conf I have:

[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;rxgain=0.0
;txgain=0.0

[interfaces]
msn=0703241432
incomingmsn=703241432
controller=1,2
softdtmf=1
accountcode=
context=default
;echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
devices=2

msn=0703241434
incomingmsn=703241434
controller=1,2
softdtmf=1
accountcode=
context=default
;echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
devices=2

Thanks,
Patrick

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RE: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

2003-12-23 Thread Sean Cheesman

The problem occurs when the software is expecting the packet in a certain
timeframe so that it can reassemble it in a timely manner.  It's not a big
deal with a web page or something along that lines.  But when a voice
application cannot get reassembled in a timely manner, you'll surely notice
it! 

-Original Message-
From: Joel Maslak
To: [EMAIL PROTECTED]
Sent: 12/23/2003 10:41 AM
Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

On Tue, 23 Dec 2003, Rich Adamson wrote:

 If a collision or dropped packet occurs (in a voip udp environment)
there
 is no way to retransmit the missing/damaged packet. Missing one packet
isn't
 a big deal, but if you have collisions and/or dropped packets, there
is a
 very high probability that lots of packets will be dropped. If too
many
 are dropped, you'll hear the result in the undecoded voice as choppy
 voice.

Actually, collisions occur at Layer 2, not Layer 3, and the layer 2
hardware automatically resends packets involved in a collision - layer 3
is never aware of it happening (although it may cause additional delay).
Eventually the ethernet card will give up if too many collisions occur
during retries, but this is very rare in practice unless the network is
*VERY* loaded.

 Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
 ethernet would handle roughly 20-25 rtp sessions before bumping into
the
 problem (your milage may vary). The majority of the folks on this list
 seem to be running home/soho systems and would likely never run into
the
 issue. But the heavier users will.

For a duplex mismatch, my experience is that if one end on a 100 Mb/sec
link is half and the other is full, bandwidth is limited to about 8
Mb/sec
max.  This is based on some tests I've accidentally conducted.  If you
try
to send 9 Mb/sec over that link, yes, some packets will get dropped as
they simply won't fit.  (But I do agree that for a half-half link, you
can
get about 20 Mb/sec)

-- 
Joel
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Re: [Asterisk-Users] abt asterisk

On Tuesday 23 December 2003 02:27, Hubert Kiyimba wrote:
 I am working on a project  vide over IP

 I am asking you to inform me whether asterisk software PBX supports
 video over IP

IAX explicitly supports images, video, and URLs.  See the gnophone
client.

-Tilghman

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AW: [Asterisk-Users] Capi Dial outgoing msn?

2003-12-23 Thread asterisk-mailing


Hi,

try it without prefix (else dtag uses first msn) -
so if your city code is 07032 and phone no (msn) 41432
- exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)


Thomas

 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Auftrag von Patrick
 Gesendet: Dienstag, 23. Dezember 2003 16:53
 An: [EMAIL PROTECTED]
 Betreff: [Asterisk-Users] Capi Dial  outgoing msn?


 Hi all,

 I am trying to get Capi Dial to use a specific outgoing msn.
 I can't get
 it to work. If I make a test call to 0703241494 (same isdn line, just
 one of the other numbers) I don't get CLID at all. Any ideas?

 ; use 0703241432 as outgoing msn
 exten = _070.,1,Dial(CAPI/@0703241432:${EXTEN}|30|r)

 in capi.conf I have:

 [general]
 nationalprefix=0
 internationalprefix=00
 rxgain=0.8
 txgain=0.8
 ;rxgain=0.0
 ;txgain=0.0

 [interfaces]
 msn=0703241432
 incomingmsn=703241432
 controller=1,2
 softdtmf=1
 accountcode=
 context=default
 ;echosquelch=1
 echocancel=yes
 echotail=64
 ;deflect=12345678
 devices=2

 msn=0703241434
 incomingmsn=703241434
 controller=1,2
 softdtmf=1
 accountcode=
 context=default
 ;echosquelch=1
 echocancel=yes
 echotail=64
 ;deflect=12345678
 devices=2

 Thanks,
 Patrick

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[Asterisk-Users] perl database get

2003-12-23 Thread Muhammad Nasim

Does anyone have any example perl agi script that does a database get. I am
being thick and can't seem to get the return value:

print DATABASE PUT big bigger biggest \n;  This bit works fine
print DATABASE GET big bigger \n;
Now what do I do to get the my value from the database get??

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Re: [Asterisk-Users] PBX Functionality How-to

2003-12-23 Thread Steven Critchfield

On Tue, 2003-12-23 at 09:48, Christopher J. Wolff wrote:
 Hello,
 
 I had a partner of mine present a Centrex 21 brochure and ask how many of
 those features can I fulfill.  There is nothing out of the ordinary, it's
 stuff like call hold, call forward, 3-way calling, etc.  Has anyone
 assembled a how-to that shows how to configure PBX or Centrex type
 functionality?  I found one in the voip-info wiki but only a couple of
 topics were filled out.

Could you at least read the documentation around here before you ask for
someone to do your work for you. If you can't be bothered to read the
documentation, at least offer to pay one of the fine consultants on the
list to do your work.  
-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

I'm not sure under what circumstances (from an overall performance 
perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
suggest choosing the size is roughly equivalent to MTU size. The 60ms
setting should result in larger packets which might be okay for high
speed uncongested links and satellite links. However, the smaller 20ms
packets effectively allow more opportunity for others to talk on the
wire and would likely improve response time for all devices on the wire.

Rich

 Interesting.  For the record, the MultiTech MVP-130 comes with a default 
 setting
 of 60ms packets on all of its supported codecs.  I changed the packet 
 sizes to
 20ms because I had never heard of anyone using such large sample sizes.
 
 Andres wrote:
 
 On Monday 22 December 2003 19:58, Rich Adamson wrote:
   
 
 On Monday 22 December 2003 16:37, Andres wrote:
   
 
 On Monday 22 December 2003 15:36, Rich Adamson wrote:
 
 
 I have a question regarding the Asterisk Packet Time for SIP Calls.
  It is hardcoded at 20ms but when I do an RTP Analysis on a stream
 it is clear that these packets are not spaced out at 20ms.  In
 general you see something like:
 
 Packet 50 - Delay 50ms
 Packet 51 - Delay 5ms
 Packet 52 - Delay 5ms
 Packet 53 - Delay 50ms
 Packet 54 - Delay 5ms
 Packet 55 - Delay 5ms
 
 Is there anyway to space them out evenly at 20ms??
 
 
 The 20 ms is not the inter-packet timing, its the relative content of
 what's within the packet. In other words, the packet contains 20ms of
 encoded voice.
 
 If the inter-packet times (delays) are large, as they would seem to
 be in your example, then something else is not right. Possibly a
 half-duplex ethernet connection, something else running on the
 server, router buffers, etc.
 
 On a typical * -- C7960 local call, I generally see from 1ms to 20ms
 inter-packet delays. Seldom (if ever) anything above 20ms.
   
 
 Thanks for your Input Rich.  I went ahead and tested this on our
 production servers and sure enough the inter-packet times are 20ms. 
 There must be something happening with our LAB Asterisk.  It could be
 the CBQ traffic shaping software we have running on it.  I will fiddle
 around with it to see if it changes anything.
 
 Thanks!
 Andres
 
 
 Ok...after some more testing, the traffic shaping software was not the
 culprit.  It turns out that if the UA is configured for 60ms of voice,
 then Asterisk will show this strange behaviour.  If we set the UA for
 20ms, then all works well.
   
 
 Cool!
 
 How did it get set to 60ms?
 
 
 The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set the 
 transmit packet size to 60ms (or multiple other values).  Asterisk will 
 receive 60ms and transmit 20ms times 3 packets, andit works quite well.  In 
 any case our SPA2000 problem was unrelated to the packet time.
 
 Regards,
 Andres 
   
 
 
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[Asterisk-Users] Re: Asterisk , Video Switching

2003-12-23 Thread Hubert Kiyimba

Dear members, 

I am writing to inquire whether Asterisk can serve as video switching 
software for the purposes of video conferencing over IP on a campus network. 

Hubert
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RE: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

There's no reassembly with udp, and there is no sense of packets arriving
in the same order as what was sent. Udp is a best-effort low-overhead way
of transmitting data (with UDP often times referred to as the Unreliable 
Data Protocol). Changing to TCP would allow reassembly, however the 
overhead would be substantial.


 The problem occurs when the software is expecting the packet in a certain
 timeframe so that it can reassemble it in a timely manner.  It's not a big
 deal with a web page or something along that lines.  But when a voice
 application cannot get reassembled in a timely manner, you'll surely notice
 it! 
 
 -Original Message-
 From: Joel Maslak
 To: [EMAIL PROTECTED]
 Sent: 12/23/2003 10:41 AM
 Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
 
 On Tue, 23 Dec 2003, Rich Adamson wrote:
 
  If a collision or dropped packet occurs (in a voip udp environment)
 there
  is no way to retransmit the missing/damaged packet. Missing one packet
 isn't
  a big deal, but if you have collisions and/or dropped packets, there
 is a
  very high probability that lots of packets will be dropped. If too
 many
  are dropped, you'll hear the result in the undecoded voice as choppy
  voice.
 
 Actually, collisions occur at Layer 2, not Layer 3, and the layer 2
 hardware automatically resends packets involved in a collision - layer 3
 is never aware of it happening (although it may cause additional delay).
 Eventually the ethernet card will give up if too many collisions occur
 during retries, but this is very rare in practice unless the network is
 *VERY* loaded.
 
  Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
  ethernet would handle roughly 20-25 rtp sessions before bumping into
 the
  problem (your milage may vary). The majority of the folks on this list
  seem to be running home/soho systems and would likely never run into
 the
  issue. But the heavier users will.
 
 For a duplex mismatch, my experience is that if one end on a 100 Mb/sec
 link is half and the other is full, bandwidth is limited to about 8
 Mb/sec
 max.  This is based on some tests I've accidentally conducted.  If you
 try
 to send 9 Mb/sec over that link, yes, some packets will get dropped as
 they simply won't fit.  (But I do agree that for a half-half link, you
 can
 get about 20 Mb/sec)
 
 -- 
 Joel
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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

2003-12-23 Thread Andres

On Tuesday 23 December 2003 10:59, Rich Adamson wrote:
 I'm not sure under what circumstances (from an overall performance
 perspective) 20ms is better then 60ms, or the reverse. Gut feeling would
In our network we set UAs to use 60ms (using G729).  Actual data measurements 
indicate a call consumes about 13Kbps.  If we use 20ms then it consumes about 
25Kbps.  These are of course peer-peer calls since Asterisk itself does not 
support transmitting at 60ms.  We prefer 60ms due to the fact that some of 
our customers are using dial-up for their VoIP, and bigger delays are 
preferable to dopped packets.

Andres.

 suggest choosing the size is roughly equivalent to MTU size. The 60ms
 setting should result in larger packets which might be okay for high
 speed uncongested links and satellite links. However, the smaller 20ms
 packets effectively allow more opportunity for others to talk on the
 wire and would likely improve response time for all devices on the wire.

 Rich
 

  Interesting.  For the record, the MultiTech MVP-130 comes with a default
  setting
  of 60ms packets on all of its supported codecs.  I changed the packet
  sizes to
  20ms because I had never heard of anyone using such large sample sizes.
 
  Andres wrote:
  On Monday 22 December 2003 19:58, Rich Adamson wrote:
  On Monday 22 December 2003 16:37, Andres wrote:
  On Monday 22 December 2003 15:36, Rich Adamson wrote:
  I have a question regarding the Asterisk Packet Time for SIP Calls.
   It is hardcoded at 20ms but when I do an RTP Analysis on a stream
  it is clear that these packets are not spaced out at 20ms.  In
  general you see something like:
  
  Packet 50 - Delay 50ms
  Packet 51 - Delay 5ms
  Packet 52 - Delay 5ms
  Packet 53 - Delay 50ms
  Packet 54 - Delay 5ms
  Packet 55 - Delay 5ms
  
  Is there anyway to space them out evenly at 20ms??
  
  The 20 ms is not the inter-packet timing, its the relative content
   of what's within the packet. In other words, the packet contains
   20ms of encoded voice.
  
  If the inter-packet times (delays) are large, as they would seem to
  be in your example, then something else is not right. Possibly a
  half-duplex ethernet connection, something else running on the
  server, router buffers, etc.
  
  On a typical * -- C7960 local call, I generally see from 1ms to
   20ms inter-packet delays. Seldom (if ever) anything above 20ms.
  
  Thanks for your Input Rich.  I went ahead and tested this on our
  production servers and sure enough the inter-packet times are 20ms.
  There must be something happening with our LAB Asterisk.  It could be
  the CBQ traffic shaping software we have running on it.  I will
   fiddle around with it to see if it changes anything.
  
  Thanks!
  Andres
  
  Ok...after some more testing, the traffic shaping software was not the
  culprit.  It turns out that if the UA is configured for 60ms of voice,
  then Asterisk will show this strange behaviour.  If we set the UA for
  20ms, then all works well.
  
  Cool!
  
  How did it get set to 60ms?
  
  The GS Phone, ATA186, and SPA2000 all have a parameter that lets you set
   the transmit packet size to 60ms (or multiple other values).  Asterisk
   will receive 60ms and transmit 20ms times 3 packets, andit works quite
   well.  In any case our SPA2000 problem was unrelated to the packet
   time.
  
  Regards,
  Andres
  
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[Asterisk-Users] turning off IAX registration attempts

2003-12-23 Thread Robert Hajime Lanning

I have, in iax.conf the register statement:
register = username:[EMAIL PROTECTED]

This causes registration attempts to iaxtel.com for both IAX and IAX2.

Every once in a while there is a packet for port 4569 keeping the IAX2
registration alive.  This is fine.

But, I have a barrage of registration attempts to iaxtel on port 5036 for
IAX.  Every UDP packet is answered with an ICMP packet claiming
port unreachable.

I know that iaxtel has turned off IAX,  So, how do I turn off the registration
attempts for IAX, for that particular connection?  (and keep IAX2)

Just seems like alot of wasted bandwidth, contiously knocking on a locked door.
Ok, not alot of bandwidth, but, completely useless.

Has anyone done a tcpdump at iaxtel to see how many IAX registration attempts
hit them, and how fast?

Here is my tcpdump: there are ICMP return packets for each of these UDP packets

[EMAIL PROTECTED]:/etc/asterisk# tcpdump -n ip host 69.73.19.178 and udp port 5036
tcpdump: listening on eth0
17:10:01.740865 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:01.740912 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:01.760869 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:01.760909 198.144.196.118.5036  69.73.19.178.5036: udp 42 (DF) [tos 0x10]
17:10:09.740652 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:11.201240 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:11.750502 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:11.750535 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:11.750546 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:11.770504 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:12.220512 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:25.240316 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:26.250264 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:29.740007 198.144.196.118.5036  69.73.19.178.5036: udp 42 (DF) [tos 0x10]
17:10:31.759849 198.144.196.118.5036  69.73.19.178.5036: udp 42 (DF) [tos 0x10]
17:10:39.279658 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:39.749612 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:40.299550 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:41.759498 198.144.196.118.5036  69.73.19.178.5036: udp 12 (DF) [tos 0x10]
17:10:41.759546 198.144.196.118.5036  69.73.19.178.5036: udp 42 (DF) [tos 0x10]

20 packets received by filter
0 packets dropped by kernel


-- 
END OF LINE
   -MCP
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Re: [Asterisk-Users] Authentication

2003-12-23 Thread Robert Mann




You have not covered very much of the 
configuration that can be done here. So with that I have come up with a 
very generic config for you that I have not tested and is to the best of my 
memory but I will give it to you as a starting point. I am posting the 
extensions.conf, zapata.conf and voicemail.conf.

It may make sense it may not. I hope 
it at least helps and does not hinder.

Assuming FXO are channels 1 and 
2
Assuming FXS are channels3 through 
5 

Since you do not have a direct mapping 
between users and extensions I gave users 1-3 direct access to Zap/3-5 and User 
4 gets stuck with a voicemail only extension.

You did not mention if you wanted a menu 
system for incoming calls so I did not create one. Instead all incoming 
calls from either line will just ring all three extensions. If no one 
picks up it goes to a generic voicemail box of 1000.

User 1 can dial 9 1234 ??? etc with 
1234 being the password for user 1

User2 can dial 9 2345 ??? etc 
with 1234 being the password for user 2
etc...

Now in reality it would probably be a 
cleaner and nicer config using the Authentication app that is available to you 
but you asked for the user to be able to just dial 9 1-4 phone number. I 
chose 4 digit passwords. If you modify that make sure you modify the 
${EXTEN:5} to what is needed. the :5 is trimming off the 9 and 4 additional 
digits for the password so if you were using 2 digit passwords you would want to 
change that to a :3.

Voicemail for each user is mute at this 
point as you have no menu system to direct a caller to a specific user hence 
voicemail here will be interoffice only at this point until you create a menu 
system or direct incoming lines to a specific user.

Use this at your own risk. I did not 
try this configuration on any box. I did this from memory and copying and 
tweaking some of my configs and my memory basically sucks so take that as you 
will. Most of what I know came from samples around the net so you will see 
a lot of stuff from various people around the internet. I am a newbie at 
this as well and did not see anyone replying to your message so I thought I 
would give it a shot at least to get you going in the right 
direction.

I am sure I forgot a lot of stuff that you 
would need but hopefully I covered what you asked for at least.

* NO FLAMES * NO FLAMES * NO FLAMES * NO 
FLAMES * NO FLAMES * NO FLAMES * NO FLAMES * NO FLAMES * NO FLAMES * 

I know I use a lot of whitespace and have 
been told numerous times not too but my system works as it is supposed 
to
so I guess I have the whitespaces in the 
proper area. Too bad if it uses more bandwidth here it makes it easier 
for
my brain to understand so you will just 
have to live with it all. If you don't like the whitespace then don't read 
the
email.

Good luck and happy holidays,

Robert

___

;zapata.conf; Channels definitions for 
zapata.conf file[channels]

language 
= en

; FXO 
Channelssignalling 
= fxs_ks ; Assuming you are using KewlStart if not change this to what you 
use.group 
= 
1callgroup 
= 
1pickupgroup 
= 
1usecallerid 
= 
yescallerid 
= 
asreceivedhidecallerid 
= 
nocallwaiting 
= yescallwaitingcallerid = 
yesthreewaycalling = 
yestransfer 
= yescancallforward = 
yesechocancel 
= yesechocancelwhenbridged = 
yesrxgain 
= 
0.6txgain 
= 
0.6immediate 
= 
nobusydetect 
= 
nocallprogress 
= 
nomusiconhold 
= random; Analog phone line attached to: 
???-???-context 
= 
fxo-line1-inmailbox 
= 1000 ; Not mapped to a specific user have them both go to generic vm 
1000channel 
= 1 ; X101P Card; Analog phone line attached to: 
???-???-context 
= 
fxo-line2-inmailbox 
= 1000 ; Not mapped to a specific user have them both go to generic vm 
1000channel 
= 2 ; X101P Card

; FXS 
Channelssignalling 
= fxo_ks ; Assuming you are using KewlStart if not change this to what you 
use.group 
= 
2callgroup 
= 
2pickupgroup 
= 
2callwaiting 
= yescallwaitingcallerid = 
yesthreewaycalling = 
yestransfer 
= yescancallforward = 
yesechocancel 
= yesechocancelwhenbridged = 
yesmailbox 
= 
callerid 
= "Zap 3" 
1context 
= 
fxo-outchannel 
= 3 ; TDM30B Port 
1mailbox 
= 
callerid 
= "Zap 4" 
2context 
= 
fxo-outchannel 
= 4 ; TDM30B Port 
2mailbox 
= 
callerid 
= "Zap 5" 
3context 
= 
fxo-outchannel 
= 5 ; TDM30B Port 3

;extensions.conf[globals]

[general]

static 
= yes ; These two 
lines prevent the command-line 
interfacewriteprotect 
= yes ; from 
overwriting the config file. Leave them here.

[macro-exten]exten 
   = 
s,1,Dial(${arg1}/${MACRO_EXTEN},${arg2})exten 
   = 
s,2,VoiceMail2(u${MACRO_EXTEN})exten 
   = 
s,3,Hangupexten  
  = 
s,102,VoiceMail2(b${MACRO_EXTEN})exten 
   = 
s,103,Hangup

[macro-no-exten]exten 
   = 
s,1,VoiceMail2(u${MACRO_EXTEN})exten 
   = 
s,2,Hangup

[extensions]

exten 
   = 
2000,1,Macro(exten,Zap/3,20) ; User 2000 with a 20 sec ring before

Re: [Asterisk-Users] sendmail problems



You say The server crashes  I assume you mean that Asterisk
core dumps and sendmail continues to run just fine.  If you
can send mail out of the box sendmail is confgured well
enough and I doubt the problem is there.

If you can get Asterisk to dump then what you need to do is
use a debugger to get a backtrace.  This will tell to the
line (as i line of coe) that caused the crash.  The thing to
remember to that if a program crashed it is due to t bug..
There _should_ be no way for a user through misconfiguration
to cause a core dump.  What you are looking for is a little
bit od C cde that doesn't handle some condition well.  If yu
use gdb and the bt commad you can find the line 
Asterisk was executing when it crashed.

I'd not suspect sendmail 

--- [EMAIL PROTECTED] wrote:
 Hello,
 
 I'm having some * and sendmail integration problems, probably because
 i don't know too much about sendmail.  My server crashes when I
 forward voicemail from one * voicemail box to another, everything
 else works.  E-mail notification works on all boxes when new mail
 arives, the problem only seems to occur during this forwarding
 function.  It's a difficult problem to troubleshoot.  If I start *
 -gc, the server doesn't crash, just hangs up for about 60 seconds
 then completes the task, so i can't seem to get a core dump to dive
 into the specifics of what's going on.  I'm not sure how to debug
 sendmail to look at that side.  If someone would be kind enough to
 e-mail me some sample sendmail.cf files, I may be able to see if I'm
 not configure properly.  I've been reading the sendmail.org site but
 this application is really archain and difficult for me to understand
 enough to fix it myself.  Thanks in advance.
 
 JR
 
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Re: [Asterisk-Users] Re: Asterisk , Video Switching

2003-12-23 Thread C. Maj

On Tue, 23 Dec 2003, Hubert Kiyimba waxed:

 Dear members, 
 
 I am writing to inquire whether Asterisk can serve as video switching 
 software for the purposes of video conferencing over IP on a campus network. 
 
 Hubert

http://www.gnophone.com/


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Re: [Asterisk-Users] PBX Functionality How-to


One thing Centrex is that Asterisk is not is a turn key
system.  With Asterisk you have to either build the PBX your
self or pay someone to build yu one.  With Centrex you
simply write a check.

THat said, you can build  anything you want so of cource the
feature list can match.

The best way to learn what it can do is to build a small PBX
with just a couple extensions.  Try to build in all the
funtionality you need in your larger system.  

If you get into trouble you may want to ask __specific__ questions
like I want to make XXX work, I triedd XXX and YYY but I still
have this problem it it?   You may have to post 50 questions
like that one at a time.  But you will get answers.  Asterisk has
a learning curve.  expect it to take a few weeks of study

But the bottom line is that Asterisk will do quite a bit more
than Centrex.  I don't think Centrex does VOIP at all


--- Christopher J. Wolff [EMAIL PROTECTED] wrote:
 Hello,
 
 I had a partner of mine present a Centrex 21 brochure and ask how
 many of
 those features can I fulfill.  There is nothing out of the ordinary,
 it's
 stuff like call hold, call forward, 3-way calling, etc.  Has anyone
 assembled a how-to that shows how to configure PBX or Centrex type
 functionality?  I found one in the voip-info wiki but only a couple
 of
 topics were filled out.
 
 Regards,
 Christopher J. Wolff, VP CIO
 Broadband Laboratories, Inc.
 http://www.bblabs.com
 
 
 
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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

2003-12-23 Thread Andres

On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
 There's no reassembly with udp, and there is no sense of packets arriving
 in the same order as what was sent. Udp is a best-effort low-overhead way
Right, UDP itself does not care about order, but at the application layer you 
can keep track of it.  You can design your application to buffer X packets 
and then reorder them according to sequence numbers.

 of transmitting data (with UDP often times referred to as the Unreliable
 Data Protocol). Changing to TCP would allow reassembly, however the
 overhead would be substantial.

 

  The problem occurs when the software is expecting the packet in a certain
  timeframe so that it can reassemble it in a timely manner.  It's not a
  big deal with a web page or something along that lines.  But when a voice
  application cannot get reassembled in a timely manner, you'll surely
  notice it!
 
  -Original Message-
  From: Joel Maslak
  To: [EMAIL PROTECTED]
  Sent: 12/23/2003 10:41 AM
  Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
 
  On Tue, 23 Dec 2003, Rich Adamson wrote:
   If a collision or dropped packet occurs (in a voip udp environment)
 
  there
 
   is no way to retransmit the missing/damaged packet. Missing one packet
 
  isn't
 
   a big deal, but if you have collisions and/or dropped packets, there
 
  is a
 
   very high probability that lots of packets will be dropped. If too
 
  many
 
   are dropped, you'll hear the result in the undecoded voice as choppy
   voice.
 
  Actually, collisions occur at Layer 2, not Layer 3, and the layer 2
  hardware automatically resends packets involved in a collision - layer 3
  is never aware of it happening (although it may cause additional delay).
  Eventually the ethernet card will give up if too many collisions occur
  during retries, but this is very rare in practice unless the network is
  *VERY* loaded.
 
   Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex 10 meg
   ethernet would handle roughly 20-25 rtp sessions before bumping into
 
  the
 
   problem (your milage may vary). The majority of the folks on this list
   seem to be running home/soho systems and would likely never run into
 
  the
 
   issue. But the heavier users will.
 
  For a duplex mismatch, my experience is that if one end on a 100 Mb/sec
  link is half and the other is full, bandwidth is limited to about 8
  Mb/sec
  max.  This is based on some tests I've accidentally conducted.  If you
  try
  to send 9 Mb/sec over that link, yes, some packets will get dropped as
  they simply won't fit.  (But I do agree that for a half-half link, you
  can
  get about 20 Mb/sec)
 
  --
  Joel
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 ---End of Original Message-


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[Asterisk-Users] WEBMIN module for Asterisk

2003-12-23 Thread Doug Shubert

Hello,
has anyone come across a module for WEBMIN to configure * ?
webmin info http://www.webmin.com/
Thanks
Doug

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Re: [Asterisk-Users] Auto Starting Asterisk



Look in the directory /etc/init.d (/etc/rc.d/init.d on
some systems)

You put a script in there called asterisk.  There is a
sample called asterisk.init in the source.  copy it to
/etc/init.d/asterisk

You may want to study the other files in /etc/init.d to see
how they work. 

Next read the chkconfig man page  and  see way you'd want to
type chkconfig --add asterisk; chkconfig asterisl on

Finally to start asterisk you can type ./asterisk start
You may also want to re-boot the computer to verify that
asterisk does start automatically


--- [EMAIL PROTECTED] wrote:
 On Tue, Dec 23, 2003 at 12:18:10PM +, Adthrawn wrote:
  Hi,
  
  Can anybody guide me in configuring the system to start Asterisk
 from 
  bootup... Probably a highly remedial question - but you've got to
 start 
  somewhere!
 
 If you use screen(1), you can do screen -d -m to start asterisk, and
 able to
 reattach to to it using screen -d -r.
 
 A sample would be like 
 
 screen -d -m /path/to/asterisk -vgc 
 
  
  Regards,
  Ad.
  
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[Asterisk-Users] OT: SIP vs. Skinny protocol

2003-12-23 Thread Peter Pauly

I assume there are several people on this list that
have Cisco Call Manager implementations under their
belt

We are beginning a call manager implementation and
the first question I asked Cisco was, should we use
SIP or Skinny. Cisco is pushing me towards Skinny, 
saying that I will lose some functionality with SIP.
They also say that most of their customers implement
skinny.

I see two obvious benefits to using SIP: 

1. I can get cheaper phones that run SIP, altough
Cisco just came out with a 7902G for $130 US. 

2. It's an open protocol and is more likely to 
survive long-term. 

What functionality do I lose by going with Skinny?

Will Cisco eventually go with SIP only and I'll have
to convert anyway?

Any other pluses or minuses?
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Re: AW: [Asterisk-Users] Capi Dial outgoing msn?

On Tue, 2003-12-23 at 17:13, [EMAIL PROTECTED] wrote:
 Hi,
 
 try it without prefix (else dtag uses first msn) -
 so if your city code is 07032 and phone no (msn) 41432
 - exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
 
 
 Thomas

Thanks for the pointer Thomas. I removed the areacode from msn= in
capi.conf and from the dial statement. Tried again and till no CLID.
Stumped at this point. Perhaps my telco doesn't allow setting outgoing
msn numbers.

Regards,
Patrick
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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)


The reason you use UDP over TCP for realtime meadia is that
TCP's ability to reliably deliver every packet in order actually
sounds worse.  Reason being is that with a UDP system a dropped
packet sounds like just a dropout but if you used TCP the audio
stream would be held up and delayed in a queue while that lost
packet was being retransmitted.  In stead of a dropout the audio
would sound as if someone kept hitts a pause button on a tape
recorder.  A dropout sounds better then a delay of potentialy 
several seconds

Almost all realtime meadia systems (telephony, video, possition
reporting and so on) maintain some kind of a buffer on the recieving
end.  But you trad the buffer lenght for delay.  Using UDP allows
the application to do the buffering where as TCP putting this buffing
functin in the operaing systems network code.



--- Andres [EMAIL PROTECTED] wrote:
 On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
  There's no reassembly with udp, and there is no sense of packets
 arriving
  in the same order as what was sent. Udp is a best-effort
 low-overhead way
 Right, UDP itself does not care about order, but at the application
 layer you 
 can keep track of it.  You can design your application to buffer X
 packets 
 and then reorder them according to sequence numbers.
 
  of transmitting data (with UDP often times referred to as the
 Unreliable
  Data Protocol). Changing to TCP would allow reassembly, however the
  overhead would be substantial.
 
  
 
   The problem occurs when the software is expecting the packet in a
 certain
   timeframe so that it can reassemble it in a timely manner.  It's
 not a
   big deal with a web page or something along that lines.  But when
 a voice
   application cannot get reassembled in a timely manner, you'll
 surely
   notice it!
  
   -Original Message-
   From: Joel Maslak
   To: [EMAIL PROTECTED]
   Sent: 12/23/2003 10:41 AM
   Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
  
   On Tue, 23 Dec 2003, Rich Adamson wrote:
If a collision or dropped packet occurs (in a voip udp
 environment)
  
   there
  
is no way to retransmit the missing/damaged packet. Missing one
 packet
  
   isn't
  
a big deal, but if you have collisions and/or dropped packets,
 there
  
   is a
  
very high probability that lots of packets will be dropped. If
 too
  
   many
  
are dropped, you'll hear the result in the undecoded voice as
 choppy
voice.
  
   Actually, collisions occur at Layer 2, not Layer 3, and the layer
 2
   hardware automatically resends packets involved in a collision -
 layer 3
   is never aware of it happening (although it may cause additional
 delay).
   Eventually the ethernet card will give up if too many collisions
 occur
   during retries, but this is very rare in practice unless the
 network is
   *VERY* loaded.
  
Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex
 10 meg
ethernet would handle roughly 20-25 rtp sessions before bumping
 into
  
   the
  
problem (your milage may vary). The majority of the folks on
 this list
seem to be running home/soho systems and would likely never run
 into
  
   the
  
issue. But the heavier users will.
  
   For a duplex mismatch, my experience is that if one end on a 100
 Mb/sec
   link is half and the other is full, bandwidth is limited to about
 8
   Mb/sec
   max.  This is based on some tests I've accidentally conducted. 
 If you
   try
   to send 9 Mb/sec over that link, yes, some packets will get
 dropped as
   they simply won't fit.  (But I do agree that for a half-half
 link, you
   can
   get about 20 Mb/sec)
  
   --
   Joel
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  ---End of Original Message-
 
 
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Re: [Asterisk-Users] perl database get

On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote:
 Does anyone have any example perl agi script that does a database
 get. I am being thick and can't seem to get the return value:

 print DATABASE PUT big bigger biggest \n;  This bit works
 fine print DATABASE GET big bigger \n;
 Now what do I do to get the my value from the database get??

$result = ;
or more explicitly:
$result = STDIN;

The real answer, though, is to point you to the Perl module at
http://asterisk.gnuinter.net and tell you that all of these issues
have already been solved.

-Tilghman

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Re: [Asterisk-Users] Music On Hold in Conference room?

2003-12-23 Thread Philipp von Klitzing

Hi!

 Does anyone here know how I might provide music into a conference room
 when there is only one participant. Dead silence tends to confuse
 non-techies who think that they've done something wrong, even after the
 entry announcement.

MeetMe() now has an option M that does exactly that. Be sure to have 
configured music-on-hold (MOH) on your system.

http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe

Cheers, Philipp


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RE: [Asterisk-Users] Asterisk + CRM

2003-12-23 Thread Michael Devenijn

Which events did you add ?
 
 


Van: Jonathan Tew [mailto:[EMAIL PROTECTED] 
Verzonden:   di 23/12/2003 16:25
Aan: [EMAIL PROTECTED]  
Onderwerp:   Re: [Asterisk-Users] Asterisk + CRM


We're starting to integrate * with our customer service software. 
Basically we're pulling off events from the management interface.  We're
also making some small patches to the code to deliver more events about
the channel variables, etc.

Anton Yurchenko wrote:

 Hello,

 Anyone aware of any CRM products projects that intagrete with *? Or
 that integrate with any telephony products? Is there some open API for
 such integration, or are they all proprietory?

 Thanks



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winmail.dat

Re: [Asterisk-Users] codes/grandstream/PRI.. few questions :)

2003-12-23 Thread Peter Brown

Justin,

Comments inline:

At 01:06 24/12/03 +1100, you wrote:
Hi Guys..

Just wondering if someone could help me with a few questions please. were 
currently using the ulaw codec with our grandstream/iconnect/asterisk 
setup and its working pretty good except for the fact it downloads heaps. 
Does anyone know a good site to get referances to how much each codec 
downloads/quality etc etc ? Ive tried using that g723 codec but i have 
have problems as soon as a i dial.

my next question.. :) does anyone know howto fix the grandstream 484 
errors you get sometimes when you dial ? i had a look at they rekon to put 
early dial on.. which just makes things worse heh.
They'd be a cool little phone except for this problem.

Lastly were looking at getting a PRI or something to handle 30 lines.. I 
know digium sells hardware to do this, has anyone in australia gotten good 
results from doing this kind of setup ?? also what are the restrictions in 
regards to caller id and that sort of stuff in aus? do is all work ?
Haven't tested GS yet so I can't help you yet.
There are a number using PRI here. We are awaiting final testing documents 
to be able to issue A Tick on TE410P-A. Have 'A tick' stickers ready to go 
to make official for Australia.
You get caller id if the number isn't silent or hasn't requested a block on 
sending the caller id.



thanks heaps everyone :)

Merry Christmas

Justin
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CEO
IP Telephonics 

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Re: [Asterisk-Users] perl database get

2003-12-23 Thread Muhammad Nasim

I've used both the syntax you have given and the perl module. AGI-getvar()
returns nothing for arguments that work from the CLI

(Also when I run agi-test.agi, the only thing that works is the SAY NUMBER.
SEND TEXT doesn't work and nothing at all is printed to teh console)

I am using redhat 8. Could it be a redhat 8 problem do you think?

- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 6:07 PM
Subject: Re: [Asterisk-Users] perl database get


 On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote:
  Does anyone have any example perl agi script that does a database
  get. I am being thick and can't seem to get the return value:
 
  print DATABASE PUT big bigger biggest \n;  This bit works
  fine print DATABASE GET big bigger \n;
  Now what do I do to get the my value from the database get??

 $result = ;
 or more explicitly:
 $result = STDIN;

 The real answer, though, is to point you to the Perl module at
 http://asterisk.gnuinter.net and tell you that all of these issues
 have already been solved.

 -Tilghman

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Re: AW: [Asterisk-Users] Capi Dial outgoing msn?

2003-12-23 Thread Philipp von Klitzing

Hi!

  try it without prefix (else dtag uses first msn) -
  so if your city code is 07032 and phone no (msn) 41432
  - exten = _070.,1,Dial(CAPI/@41432:${EXTEN}|30|r)
  
  
  Thomas
 
 Thanks for the pointer Thomas. I removed the areacode from msn= in
 capi.conf and from the dial statement. Tried again and till no CLID.
 Stumped at this point. Perhaps my telco doesn't allow setting outgoing
 msn numbers.

If you read the CAPI documentation you'll find that @ will help you to 
_hide_ your ID (this is called CLIR) - however from your message I 
understand that you want to do the opposite? So just drop the @.

If your problem persists: You might have told your telco to _always_ hide 
your ID. Or maybe it's just that you need to remove the 0 before 
703241432 as outgoing MSN.

Cheers, Philipp


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Fw: [Asterisk-Users] perl database get

2003-12-23 Thread Muhammad Nasim

i mean AGI-database_get()

- Original Message -
From: Muhammad Nasim [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 6:41 PM
Subject: Re: [Asterisk-Users] perl database get

 I've used both the syntax you have given and the perl module.
AGI-getvar()
 returns nothing for arguments that work from the CLI

 (Also when I run agi-test.agi, the only thing that works is the SAY
NUMBER.
 SEND TEXT doesn't work and nothing at all is printed to teh console)

 I am using redhat 8. Could it be a redhat 8 problem do you think?

 - Original Message -
 From: Tilghman Lesher [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, December 23, 2003 6:07 PM
 Subject: Re: [Asterisk-Users] perl database get

  On Tuesday 23 December 2003 10:20, Muhammad Nasim wrote:
   Does anyone have any example perl agi script that does a database
   get. I am being thick and can't seem to get the return value:

   print DATABASE PUT big bigger biggest \n;  This bit works
   fine print DATABASE GET big bigger \n;
   Now what do I do to get the my value from the database get??

  $result = ;
  or more explicitly:
  $result = STDIN;

  The real answer, though, is to point you to the Perl module at
  http://asterisk.gnuinter.net and tell you that all of these issues
  have already been solved.

  -Tilghman

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Re: [Asterisk-Users] turning off IAX registration attempts

On Tuesday 23 December 2003 11:13, Robert Hajime Lanning wrote:
 I have, in iax.conf the register statement:
 register = username:[EMAIL PROTECTED]

 This causes registration attempts to iaxtel.com for both IAX and
 IAX2.

 Every once in a while there is a packet for port 4569 keeping the
 IAX2 registration alive.  This is fine.

 But, I have a barrage of registration attempts to iaxtel on port
 5036 for IAX.  Every UDP packet is answered with an ICMP packet
 claiming port unreachable.

 I know that iaxtel has turned off IAX,  So, how do I turn off the
 registration attempts for IAX, for that particular connection? 
 (and keep IAX2)

How's this for a solution (attached)?

-Tilghman
Index: channels/chan_iax.c
===
RCS file: /usr/cvsroot/asterisk/channels/chan_iax.c,v
retrieving revision 1.43
diff -u -r1.43 chan_iax.c
--- channels/chan_iax.c	9 Dec 2003 23:55:17 -	1.43
+++ channels/chan_iax.c	23 Dec 2003 18:43:41 -
@@ -4661,7 +4661,7 @@
 } else if (!strcasecmp(v-value, yes)) {
 	peer-maxms = DEFAULT_MAXMS;
 } else if (sscanf(v-value, %d, peer-maxms) != 1) {
-	ast_log(LOG_WARNING, Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of iax.conf\n, peer-name, v-lineno);
+	ast_log(LOG_WARNING, Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of iax1.conf\n, peer-name, v-lineno);
 	peer-maxms = 0;
 }
 			} //else if (strcasecmp(v-name,type))
@@ -4962,7 +4962,7 @@
 
 static int reload_config(void)
 {
-	char *config = iax.conf;
+	char *config = iax1.conf;
 	struct iax_registry *reg;
 	struct sockaddr_in dead_sin;
 	strncpy(accountcode, , sizeof(accountcode)-1);
@@ -5359,7 +5359,7 @@
 
 int load_module(void)
 {
-	char *config = iax.conf;
+	char *config = iax1.conf;
 	int res = 0;
 	int x;
 	struct iax_registry *reg;

[Asterisk-Users] Cisco 7960 phones.

2003-12-23 Thread Ariel Batista

I have found a phone that I wish I had not!  This is by far the worst
phone to setup.  I have finally upgraded it to Sip but once this got
done it I am not able to get it unlocked so I can enter the rest of the
settings.  So if anyone out there can tell me how to setup my DNS server
to tell it where the tftp is located (Windows 2000 DNS server) or please
let me have the default password.  I called Cisco and they said I can
change it with the tftp setup and that is all they said. (They also said
to use Skinny and not SIP) Not much help from there support contract we
have on with them!  Why have they made a phone that is so hard to get
working!  I feel that Cisco does not understand the KISS format!  I have
read the Wiki and the setup is listed there incorrectly as 9740/9760
instead of 7940/7960 (Can someone rename that please).

Thank you all for allowing to do a little venting as well.  Sorry did
not mean to do this!  But this has gotten me to loose hair and I don't
have much left!

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[Asterisk-Users] please help - ztdummy problems

2003-12-23 Thread Hector Q.-datafull

I have read a lot about ztdummy, but I miss something.
I don't have any digium hardware, but want to do Meetme.
I read that I need ztdummy installed in order to do a conference room.
I followed all steps to get ztdummy compiled and installed (including uncoment on 
makefile)
When I install the module, my * sound becomes unrecognizably choppy on every channel 
type,
not only meetme!
So I have been forced to uninstall the module to return my * to an usable state.
Anybody can help me to solve this?
Thanks a lot.
Hector.

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Re: AW: [Asterisk-Users] Capi Dial outgoing msn?

On Tue, 2003-12-23 at 19:39, Philipp von Klitzing wrote:
[snip]
 If you read the CAPI documentation you'll find that @ will help you to 
 _hide_ your ID (this is called CLIR) - however from your message I 
 understand that you want to do the opposite? So just drop the @.
 
 If your problem persists: You might have told your telco to _always_ hide 
 your ID. Or maybe it's just that you need to remove the 0 before 
 703241432 as outgoing MSN.
 
 Cheers, Philipp
 

Hi Philipp,

The text in the chan_capi README sort of tells you to use it:

Using CLIR
 ==
in the Dial command put a '@' infront of the msn you want to use for
dialing out, e.g.:
s,1,Dial,CAPI/@12345678:BYEXTENSION|30|r

My interpretation of that text is that I need to use the @ so I can
set the outgoing MSN. If I had known that the R in CLIR means something
like Restrict it would have been obvious :)

Luckily your pointer solved my problem so thanks for that.

Regards,
Patrick

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Re: [Asterisk-Users] Cisco 7960 phones.

What firmware did you upgrade to?

If its version 5.0 and above the default password is cisco and to unlock
it you press settings then 9.

NO cisco's docs are simple.. You are just trying too hard.

bkw

On Tue, 23 Dec 2003, Ariel Batista wrote:

 I have found a phone that I wish I had not!  This is by far the worst
 phone to setup.  I have finally upgraded it to Sip but once this got
 done it I am not able to get it unlocked so I can enter the rest of the
 settings.  So if anyone out there can tell me how to setup my DNS server
 to tell it where the tftp is located (Windows 2000 DNS server) or please
 let me have the default password.  I called Cisco and they said I can
 change it with the tftp setup and that is all they said. (They also said
 to use Skinny and not SIP) Not much help from there support contract we
 have on with them!  Why have they made a phone that is so hard to get
 working!  I feel that Cisco does not understand the KISS format!  I have
 read the Wiki and the setup is listed there incorrectly as 9740/9760
 instead of 7940/7960 (Can someone rename that please).

 Thank you all for allowing to do a little venting as well.  Sorry did
 not mean to do this!  But this has gotten me to loose hair and I don't
 have much left!

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[Asterisk-Users] Voiceglo setup for home

2003-12-23 Thread Cameron Palmer

I am looking to speak to anyone else that has connected to Voiceglo using 
Asterisk. I'm using SIP and have most of the issues worked out. But remote 
outbound ringing doesn't work. So it would be nice to discuss configs. 
Maybe someone out there is using IAX instead.

cameron.



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Re: [Asterisk-Users] Cisco 7960 phones.

2003-12-23 Thread Ariel Batista

Brian West wrote:
 What firmware did you upgrade to?

 If its version 5.0 and above the default password is cisco and to
 unlock it you press settings then 9.

 NO cisco's docs are simple.. You are just trying too hard.

I want to thank you for the password of cisco.  It worked. I have
finally gotten the phone to work.  Now to start setting all the new
bells and tones it has!  I still think that they have over done the
settings on the phone!

 bkw

 On Tue, 23 Dec 2003, Ariel Batista wrote:

 I have found a phone that I wish I had not!  This is by far the worst
 phone to setup.  I have finally upgraded it to Sip but once this got
 done it I am not able to get it unlocked so I can enter the rest of
 the settings.  So if anyone out there can tell me how to setup my
 DNS server to tell it where the tftp is located (Windows 2000 DNS
 server) or please let me have the default password.  I called Cisco
 and they said I can change it with the tftp setup and that is all
 they said. (They also said to use Skinny and not SIP) Not much help
 from there support contract we have on with them!  Why have they
 made a phone that is so hard to get working!  I feel that Cisco does
 not understand the KISS format!  I have read the Wiki and the setup
 is listed there incorrectly as 9740/9760 instead of 7940/7960 (Can
 someone rename that please).

 Thank you all for allowing to do a little venting as well.  Sorry did
 not mean to do this!  But this has gotten me to loose hair and I
 don't have much left!

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Re: [Asterisk-Users] Cisco 7960 phones.

2003-12-23 Thread Cameron Palmer

I've got to agree. Once you figure out the first phone, all the others
take about 30 seconds to configure.

The Cisco SIP documentation is located at:
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/sip/index.htm

cameron.

On Tue, 23 Dec 2003, Brian West wrote:

What firmware did you upgrade to?

If its version 5.0 and above the default password is cisco and to unlock
it you press settings then 9.

NO cisco's docs are simple.. You are just trying too hard.

bkw

On Tue, 23 Dec 2003, Ariel Batista wrote:

I have found a phone that I wish I had not! This is by far the worst
phone to setup. I have finally upgraded it to Sip but once this got
done it I am not able to get it unlocked so I can enter the rest of the
settings. So if anyone out there can tell me how to setup my DNS server
to tell it where the tftp is located (Windows 2000 DNS server) or please
let me have the default password. I called Cisco and they said I can
change it with the tftp setup and that is all they said. (They also said
to use Skinny and not SIP) Not much help from there support contract we
have on with them! Why have they made a phone that is so hard to get
working! I feel that Cisco does not understand the KISS format! I have
read the Wiki and the setup is listed there incorrectly as 9740/9760
instead of 7940/7960 (Can someone rename that please).

Thank you all for allowing to do a little venting as well. Sorry did
not mean to do this! But this has gotten me to loose hair and I don't
have much left!

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Re: [Asterisk-Users] perl database get

On Tuesday 23 December 2003 12:41, Muhammad Nasim wrote:
 I've used both the syntax you have given and the perl module.
 AGI-getvar() returns nothing for arguments that work from the CLI

Try AGI-get_variable()

 (Also when I run agi-test.agi, the only thing that works is the SAY
 NUMBER. SEND TEXT doesn't work and nothing at all is printed to teh
 console)

SEND TEXT does not print anything on the console.  SEND TEXT sends
text on the channel, if the channel supports it.  Audio-only channels
do not support the SEND TEXT command.

-Tilghman

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Re: Fw: [Asterisk-Users] perl database get

On Tuesday 23 December 2003 12:44, Muhammad Nasim wrote:
 i mean AGI-database_get()

Then that probably means that the database key does not exist.

-Tilghman

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Fw: [Asterisk-Users] Fw: Questions and finding

2003-12-23 Thread Jess Magnaye


 Thanks for the reply.

 1. My VAD is turned off (00140014), and it didn't help for that cut-off.
I
 am not sure if OutboundProxy has to be configured to have it working fine.
 Or this just happened to me?  What is your ATA's software?

 2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833.  None
worked.
 As per ATA, it is by default using rfc2833.  I tried setting it up as
inband
 by setting Audiomode, but nothing helped.  I was thinking the * is ONLY
 recognizing the DTMF if there is telco board installed.  Is it?


 - Original Message - 
 From: Philipp von Klitzing [EMAIL PROTECTED]
 To: Jess Magnaye [EMAIL PROTECTED]
 Sent: Tuesday, December 23, 2003 12:36 PM
 Subject: Re: [Asterisk-Users] Fw: Questions and finding


  Hi!
 
   1.) First test
   - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off
   after 5-10seconds (consistently).
   - Solution: I have to reconfigure ATA to use OutboundProxy to be
 Asterisk
   IP.
   - Am I doing the right thing?
 
  Turn of silence detection / VAD.
 
   Any solution to this one?
   My thinking was that DTMF can only be detected by *
 
  Take a look at your SIP configuration and make sure you have the correct
  dtmfmode= set. Try different values if you continue to have trouble and
  configure your ATA accordingly.
 
  Cheers, Philipp
 
 


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Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

2003-12-23 Thread Lists

How do you reset the unit without pulling out the plug.  The easiest way 
to get the info you are looking for, is to get an 8 buck CCO account.


On 
Tue, 23 Dec 2003, Adthrawn wrote:

 Hi,
 
 Has anybody been successful in running the 7914 expansion unit for the 
 Cisco 7960G IP phone? For anybody unaware of what the expansion unit 
 does, it provides 14 additional buttons, with an LCD display. The idea, 
 is that with an expansion unit (a 7960 can take upto 2 of these units), 
 a user can either assign more speed-dial's, or can monitor line 
 status/account status. So, you can either register a speed-dial or 
 register another account.
 
 The problem I've found so far, is that speed-dials are not programmed 
 on the phone, but are instead handled by the Call Manager software (not 
 on a user basis, but on a phone, MAC address basis). Likewise, plugging 
 the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red 
 lights (the buttons also light-up red, blue or green), which according 
 to the hidden technical documentation, indicates that the Call Manager 
 is not registering the unit. I can't work out if it's short of firmware 
 embedded in the Call Manager, whether it's searching for a 
 configuration file on the TFTP (Cisco phones need a TFTP to get their 
 settings and SIP firmware), whether it's not happy with the phone being 
 a SIP version, or whether I'm doing something wrong.
 
 I've had to learn about the 7960's configuration the hard way, and 
 despite their useless technical documents, have managed to configure 
 most settings.
 
 There's quite a bit of extra configuration for the 7960 I'd love to get 
 to, and would like help or advice on. Things like directory services, 
 screen logo, the 7914 and more!
 
 If anybody is interested, I have resources and files to; convert from 
 Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail 
 indicator lamp, special key combinations to reset the unit (without 
 pulling the plug out) and locking/unlocking the preferences, 
 configuring the voicemail speed-dial
 
 Any help or advice, please let me know!
 
 Regards,
 Ad.
 
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Re: [Asterisk-Users] Fw: Questions and finding

2003-12-23 Thread Jess Magnaye



 Hi!

  1. My VAD is turned off (00140014), and it didn't help for that cut-off.

 Then check if you have a firewall in between * and your ATA that closes
 the port due to inactivity of your ATA. Also use SIP DEBUG in the CLI
 to try to see a bit more of what is going on. You could also use Ethereal
 to monitor the SIP traffic (or the rtp/UDP traffic).

  am not sure if OutboundProxy has to be configured to have it working
fine.
  Or this just happened to me?  What is your ATA's software?

 I don't have such a device, in fact never had. :-)

MY ATA and * are sitting on the same LAN.  So FW or NAT problem is not
possible.  This is also the reason why I commented out nat=1 in the
sip.conf.

  2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833.  None
worked.

 Note: inband only works with g.711 codec. Doesn't the ATA also offer
 info as third dtmfmode option? Anyway, you might want to search the
 mailing list for setup info, there are a lot of people around that use
 it.

  by setting Audiomode, but nothing helped.  I was thinking the * is
  ONLY recognizing the DTMF if there is telco board installed.  Is it?

 No no, * doesn't require any hardware to be installed.

LET ME TRY dtmfmode=info  AND SEE WHAT HAPPENS NEXT.

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Re: [Asterisk-Users] Auto Starting Asterisk

2003-12-23 Thread SW

Hi Chris,

In this situation, how do I modprobe ztdumy before * get started ?

SW

Message: 6
Date: Tue, 23 Dec 2003 09:33:07 -0800 (PST)
From: Chris Albertson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Auto Starting Asterisk
To: [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]

Look in the directory /etc/init.d (/etc/rc.d/init.d on
some systems)

You put a script in there called asterisk.  There is a
sample called asterisk.init in the source.  copy it to
/etc/init.d/asterisk

You may want to study the other files in /etc/init.d to see
how they work. 

Next read the chkconfig man page  and  see way you'd want to
type chkconfig --add asterisk; chkconfig asterisl on

Finally to start asterisk you can type ./asterisk start
You may also want to re-boot the computer to verify that
asterisk does start automatically

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Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

7914's don't work with SIP.  SCCP only.  And why do people keep talking
about this 8 dollar CCO account ... Its a service contract on the Cisco
ATA-186.  The one for the 79XX's are over 80.00/yr

bkw

On Tue, 23 Dec 2003, Lists wrote:

 How do you reset the unit without pulling out the plug.  The easiest way
 to get the info you are looking for, is to get an 8 buck CCO account.


 On
 Tue, 23 Dec 2003, Adthrawn wrote:

  Hi,
 
  Has anybody been successful in running the 7914 expansion unit for the
  Cisco 7960G IP phone? For anybody unaware of what the expansion unit
  does, it provides 14 additional buttons, with an LCD display. The idea,
  is that with an expansion unit (a 7960 can take upto 2 of these units),
  a user can either assign more speed-dial's, or can monitor line
  status/account status. So, you can either register a speed-dial or
  register another account.
 
  The problem I've found so far, is that speed-dials are not programmed
  on the phone, but are instead handled by the Call Manager software (not
  on a user basis, but on a phone, MAC address basis). Likewise, plugging
  the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red
  lights (the buttons also light-up red, blue or green), which according
  to the hidden technical documentation, indicates that the Call Manager
  is not registering the unit. I can't work out if it's short of firmware
  embedded in the Call Manager, whether it's searching for a
  configuration file on the TFTP (Cisco phones need a TFTP to get their
  settings and SIP firmware), whether it's not happy with the phone being
  a SIP version, or whether I'm doing something wrong.
 
  I've had to learn about the 7960's configuration the hard way, and
  despite their useless technical documents, have managed to configure
  most settings.
 
  There's quite a bit of extra configuration for the 7960 I'd love to get
  to, and would like help or advice on. Things like directory services,
  screen logo, the 7914 and more!
 
  If anybody is interested, I have resources and files to; convert from
  Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail
  indicator lamp, special key combinations to reset the unit (without
  pulling the plug out) and locking/unlocking the preferences,
  configuring the voicemail speed-dial
 
  Any help or advice, please let me know!
 
  Regards,
  Ad.
 
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[Asterisk-Users] Conf file system generation in * for User/Admin update

2003-12-23 Thread fred alexander

Is there anyone who could show me code (or point me in
the right direction) to allow users or PABX Admin to
generate their own * conf files.

If there isn't anything I will just have to start it
myself. Any suggestions for basics to start with. I
believe the issues are going to be about dependencies
between the various conf. files.

Any help would be gratefully received.

A GUI for this would be great or just via a web
browser.



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Re: [Asterisk-Users] Fw: Questions and finding

2003-12-23 Thread Jess Magnaye

Hi Philip, I found the problem.  My sip.conf config was changed by somebody
else. :(  The external IP was uncommented and that's what is causing my
problem.

- Original Message - 
From: Jess Magnaye [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 4:33 PM
Subject: Re: [Asterisk-Users] Fw: Questions and finding

  Hi!

   1. My VAD is turned off (00140014), and it didn't help for that
cut-off.

  Then check if you have a firewall in between * and your ATA that closes
  the port due to inactivity of your ATA. Also use SIP DEBUG in the CLI
  to try to see a bit more of what is going on. You could also use
Ethereal
  to monitor the SIP traffic (or the rtp/UDP traffic).

   am not sure if OutboundProxy has to be configured to have it working
 fine.
   Or this just happened to me?  What is your ATA's software?

  I don't have such a device, in fact never had. :-)

 MY ATA and * are sitting on the same LAN.  So FW or NAT problem is not
 possible.  This is also the reason why I commented out nat=1 in the
 sip.conf.

   2. I tried dtmfmode=inband on sip.conf, and dtmfmode=rfc2833.  None
 worked.

  Note: inband only works with g.711 codec. Doesn't the ATA also offer
  info as third dtmfmode option? Anyway, you might want to search the
  mailing list for setup info, there are a lot of people around that use
  it.

   by setting Audiomode, but nothing helped.  I was thinking the * is
   ONLY recognizing the DTMF if there is telco board installed.  Is it?

  No no, * doesn't require any hardware to be installed.

 LET ME TRY dtmfmode=info  AND SEE WHAT HAPPENS NEXT.

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[Asterisk-Users] Cisco 7960 Sounds patchy.

2003-12-23 Thread Ariel Batista

I have gotten the Cisco 7960 working with my Asterisk system under SIP.
The version is 5.03 that I am using.  Cisco Support said I should not
upgrade to version 6 yet. My next question is the sound is patchy when
people here me.  But I can hear them just fine not patchy.  I have the
188 page Admin manual and it seem not to say anything about improving
the sound. All other phones like IPDialog work fine without the patchy
sound.   I have tried ulaw and alaw as the codex.  Both sound the same!
Is there any other settings that can be done.  I remember that the
X-lite has a transmit silence but I could not find this setting in there
documentation.

P.S. the Contract for the 7960 cost us  $ 83.40 for each phone.  This I
feel is high.

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[Asterisk-Users] Packet8 Minus the DTA

2003-12-23 Thread Scott Bennett









I know someone mentioned doing this once before however I
cant find it.



Anyone remember if or how it was successful?



Thanks!

Re: [Asterisk-Users] Cisco 7960 Sounds patchy.

2003-12-23 Thread Jeremy McNamara

Ariel Batista wrote:

P.S. the Contract for the 7960 cost us  $ 83.40 for each phone.  This I
feel is high.
 

This smells like a Cisco re-certification fee to me.

Jeremy McNamara

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Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs

2003-12-23 Thread Balaji NJL




resending.

Can anyone help me in trying to understand what 
would be the problem. appreciate ur time. i need to get this 
working.

thanks a lot,
-B

  - Original Message - 
  From: 
  Balaji NJL 
  
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, December 22, 2003 8:15 
  PM
  Subject: [Asterisk-Users] MSN to GS - 
  Call drops in 10 secs
  
  Hi All,
  
  i dont know what changes i made recently but i am 
  unable to hold the call for more 10 secs between MSN and GS. PSTN to MSN and 
  PSTN to GS and vice versa works fine. i am not behind NAT.Also MSN to MSN 
  works fine too.
  
  my SIP details
  
  [general]port = 5060bindaddr = 
  0.0.0.0context = bogon-calls;context = 
  defaultdisallow=allallow=ulawallow=alawallow=ilbcallow=gsm
  
  ;My SIP phone - 
  GS[2000]type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband
  
  ;MSN 
  Msgr[2002]type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext
  i did a SIP trace
  
  it says Format=UKN
  CSeq=BYE
  
  thanks for the help,
  -Balaji
  
  
  Do you Yahoo!?Yahoo! Photos - Get 
  your photo on the big screen in Times Square

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[Asterisk-Users] Merry Christmas, all Asterisk users!

2003-12-23 Thread Olle E. Johansson

It's the day before Christmas here in Sweden, actually the night before at this time...

We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
merry-christmas-mode with the yet undocumented CLI command frosty-mode on, a mode
where the PBX will connect all incoming extensions to the ho-ho-ho sound file and 
then randomly
pick a number in the +1234 country code (for the North Pole), dial out and bridge. And 
these
magical SIP connections will work over ANY type of NAT.
(Due to the SIP header Santa-magic-cookie: on)
And yes, the frosty mode is even un-documented on the http://www.voip-info.org wiki. :-)

It's been fun spending the fall with the Asterisk project. I look forward to next year,
with the new handbook coming in place, with many new applications
and features and - hopefully - many new Asterisk installs at customer sites.
It's snowing outside, the trees are already covered with snow and the stars are 
glittering on
a dark sky. My kids are sleeping, dreaming of their christmas gifts tomorrow. It's 
going
to be a traditional Swedish christmas...
Have a wonderful Christmas, all of you!

Warm regards,
/Olle
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[Asterisk-Users] SIP / FXS - MOH

2003-12-23 Thread PBX

Is there anway to do MOH on a FXS extension like what is done using SIP.
There has to be a way within manager or something, to send this call to
MOH and then retreive the call.

I need to set this up, so users are just hitting one button to put
callers on hold and one or another button to retrieve the users.

-gcc
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[Asterisk-Users] configuration files for cisco 7960

2003-12-23 Thread Paul Mona









Is there any place where I can download sample files for the
cisco 7960 (SIP) ?

[Asterisk-Users] Voiceglo SIP configuration

2003-12-23 Thread Cameron Palmer

The call quality is really pretty good. I think better than Vonage over 
an FXO bridge. If you are looking for a home provider with direct SIP 
support and local phone numbers this is a good choice. If anyone has 
questions or comments about my configuration please pass them along. I 
have noticed that if you don't put fromuser=phone# then the extension 
caller id passes through. Also the major annoyance is why outbound 
calling gives no ring indication. I'm still looking into whether there is 
no ring indicator being sent back, or how to create one. Using the little 
'r' at the end of the dial string just seems to prevent the call from 
going through.


Username is your 10-digit phone number.
Password is in the .reg file they sent you via email. I signed up for the 
USB phone, so I don't if they send a .reg file if you went for the MTA.

sip.conf
register = 1234567890:[EMAIL PROTECTED]

[myphone.voiceglo.com]
type=peer
username=1234567890
secret=password
nat=no
host=myphone.voiceglo.com
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=200
restrictid=no 
fromuser=1234567890
fromdomain=1234567890.voiceglo.com

extensions.conf
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])

cameron.





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Re: [Asterisk-Users] Merry Christmas, all Asterisk users!

2003-12-23 Thread firedude

Merry Christmas Ollie from all of us Asterisk people in the US/East Coast 
region.
AJ

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RE: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

2003-12-23 Thread Paul Mahler

If you purchase a new telephone, the warranty is more like $15. It's more
for used phones. 

 
Paul Mahler 
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian West
Sent: Tuesday, December 23, 2003 2:17 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone)
 Help With 7960's Speed-dials

7914's don't work with SIP.  SCCP only.  And why do people keep talking
about this 8 dollar CCO account ... Its a service contract on the Cisco
ATA-186.  The one for the 79XX's are over 80.00/yr

bkw

On Tue, 23 Dec 2003, Lists wrote:

 How do you reset the unit without pulling out the plug.  The easiest way
 to get the info you are looking for, is to get an 8 buck CCO account.


 On
 Tue, 23 Dec 2003, Adthrawn wrote:

  Hi,
 
  Has anybody been successful in running the 7914 expansion unit for the
  Cisco 7960G IP phone? For anybody unaware of what the expansion unit
  does, it provides 14 additional buttons, with an LCD display. The idea,
  is that with an expansion unit (a 7960 can take upto 2 of these units),
  a user can either assign more speed-dial's, or can monitor line
  status/account status. So, you can either register a speed-dial or
  register another account.
 
  The problem I've found so far, is that speed-dials are not programmed
  on the phone, but are instead handled by the Call Manager software (not
  on a user basis, but on a phone, MAC address basis). Likewise, plugging
  the 7914 unit into the phone on an Asterisk PBX, just brings up 14 red
  lights (the buttons also light-up red, blue or green), which according
  to the hidden technical documentation, indicates that the Call Manager
  is not registering the unit. I can't work out if it's short of firmware
  embedded in the Call Manager, whether it's searching for a
  configuration file on the TFTP (Cisco phones need a TFTP to get their
  settings and SIP firmware), whether it's not happy with the phone being
  a SIP version, or whether I'm doing something wrong.
 
  I've had to learn about the 7960's configuration the hard way, and
  despite their useless technical documents, have managed to configure
  most settings.
 
  There's quite a bit of extra configuration for the 7960 I'd love to get
  to, and would like help or advice on. Things like directory services,
  screen logo, the 7914 and more!
 
  If anybody is interested, I have resources and files to; convert from
  Cisco Skinny/MGCP to a SIP version, how to configure the VoiceMail
  indicator lamp, special key combinations to reset the unit (without
  pulling the plug out) and locking/unlocking the preferences,
  configuring the voicemail speed-dial
 
  Any help or advice, please let me know!
 
  Regards,
  Ad.
 
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Re: [Asterisk-Users] MSN messenger and *

2003-12-23 Thread Glen

Speaking of MSN/Windows Messenger, how does one call someone?  
Using the configuration specified, I've registered it with Asterisk, but
it requires that I add a Passport contact.  

Does anyone have experience calling a sip endpoint without it being a
Passport account?  

-g


On Mon, 2003-12-22 at 20:42, Balaji NJL wrote:
 use this
  
 
 [3001]
 
 type=friend
 
 ;username=3001
 
 ;fromuser=Craig1
 
 ;secret=secret
 
 host=dynamic
 
 mailbox=3001
 
 context=sip
 
 dtmfmode=info
 
 auth=plaintext
 
  
 
 make sure ur MSN version is 4.7.0105. 
 
  
 
 -B
 
 
 - Original Message - 
 From: Craig Waddington
 To: [EMAIL PROTECTED]
 Sent: Monday, December 22, 2003 10:10 AM
 Subject: [Asterisk-Users] MSN messenger and *
 
 
 Sorry for the late reply. 
 
  
 
 I try port 5060 and it just knocks me back straight away, I
 cant see it even try to authenticate in the CLI.
 
  
 
 X-lite works both inside the LAN and outside using SIP.
 
  
 
 Messenger version = 4.7
 
  
 
 John I will try your suggestion with sip.conf thanks for the
 help. I notice a few differences, I seem to be missing some
 bits..
 
  
 
 Its like it is trying to authenticate with the Linux box and
 not asterisk.
 
  
 
 Sip.conf
 
  
 
 [general]
 
 port=5060   ; Port to bind to
 
 bindaddr=0.0.0.0; Address to bind to
 
 context=sip ; Default for incoming calls
 
 allow=ulaw
 
 allow=alaw
 
 allow=gsm
 
 allow=ilbc
 
  
 
  
 
 [3001]
 
 type=friend
 
 username=3001
 
 fromuser=Craig1
 
 secret=secret
 
 host=dynamic
 
 mailbox=3001
 
 context=sip
 
 dtmfmode=info
 
  
 
 I found 3 guides and each one seems to be a bit different and
 use different ports.
 
  
 
 I am using the X100P, it is a home system, to reduce call
 charges for my family overseas.
 
  
 
 If  I can get Messengger working it will be easier to talk
 them through the setup.
 
  
 
  
 
  
 
 
 
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RE: [Asterisk-Users] MSN to GS - Call drops in 10 secs

2003-12-23 Thread Craig Waddington









Balaji,



I also have the
same issue. Works fine on any phone except GS for me.



After a bit of
research I found a post saying set the phone to offer only one codec set.



It looks like we
have to set the phone to use one codec  GSM 



I am concerned
that you cant use passwords when logging in to * using Messenger.



Craig.













From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Balaji NJL
Sent: 23 December 2003 23:04
To:
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] MSN
to GS - Call drops in 10 secs







resending.











Can anyone help me in trying to understand what would be the
problem. appreciate ur
time. i need to get this working.











thanks a lot,





-B







- Original Message - 





From: Balaji NJL 





To: [EMAIL PROTECTED] 





Sent: Monday, December
22, 2003 8:15 PM





Subject: [Asterisk-Users]
MSN to GS - Call drops in 10 secs











Hi All,











i dont know what changes i made recently but i am unable to
hold the call for more 10 secs between MSN and GS. PSTN to MSN and PSTN to GS
and vice versa works fine. i am not behind NAT.Also MSN to MSN works fine too.











my SIP details











[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
;context = default
disallow=all
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm











;My SIP phone - GS
[2000]
type=friend
username=2000
secret=qweqwe
host=dynamic
context=from-sip
mailbox=2000
dtmfmode=inband











;MSN Msgr
[2002]
type=friend
host=dynamic
insecure=yes
dtmfmode=inband
;dtmfmode=rfc2833
context=from-sip
mailbox=2002
;auth=plaintext





i did a SIP trace











it says Format=UKN





CSeq=BYE











thanks for the help,





-Balaji









Do you Yahoo!?
Yahoo! Photos - Get
your photo on the big screen in Times Square









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Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)

100% agree. I think this thread is getting strung out much further
then Olle's original question relative to commenting on half vs full 
duplex.

Lots of great discussion though thanks to all that participated!

Rich


 The reason you use UDP over TCP for realtime meadia is that
 TCP's ability to reliably deliver every packet in order actually
 sounds worse.  Reason being is that with a UDP system a dropped
 packet sounds like just a dropout but if you used TCP the audio
 stream would be held up and delayed in a queue while that lost
 packet was being retransmitted.  In stead of a dropout the audio
 would sound as if someone kept hitts a pause button on a tape
 recorder.  A dropout sounds better then a delay of potentialy 
 several seconds
 
 Almost all realtime meadia systems (telephony, video, possition
 reporting and so on) maintain some kind of a buffer on the recieving
 end.  But you trad the buffer lenght for delay.  Using UDP allows
 the application to do the buffering where as TCP putting this buffing
 functin in the operaing systems network code.
 
 
 
 --- Andres [EMAIL PROTECTED] wrote:
  On Tuesday 23 December 2003 11:40, Rich Adamson wrote:
   There's no reassembly with udp, and there is no sense of packets
  arriving
   in the same order as what was sent. Udp is a best-effort
  low-overhead way
  Right, UDP itself does not care about order, but at the application
  layer you 
  can keep track of it.  You can design your application to buffer X
  packets 
  and then reorder them according to sequence numbers.
  
   of transmitting data (with UDP often times referred to as the
  Unreliable
   Data Protocol). Changing to TCP would allow reassembly, however the
   overhead would be substantial.
  
   
  
The problem occurs when the software is expecting the packet in a
  certain
timeframe so that it can reassemble it in a timely manner.  It's
  not a
big deal with a web page or something along that lines.  But when
  a voice
application cannot get reassembled in a timely manner, you'll
  surely
notice it!
   
-Original Message-
From: Joel Maslak
To: [EMAIL PROTECTED]
Sent: 12/23/2003 10:41 AM
Subject: Re: [Asterisk-Users] Asterisk SIP Packet Time (20ms)
   
On Tue, 23 Dec 2003, Rich Adamson wrote:
 If a collision or dropped packet occurs (in a voip udp
  environment)
   
there
   
 is no way to retransmit the missing/damaged packet. Missing one
  packet
   
isn't
   
 a big deal, but if you have collisions and/or dropped packets,
  there
   
is a
   
 very high probability that lots of packets will be dropped. If
  too
   
many
   
 are dropped, you'll hear the result in the undecoded voice as
  choppy
 voice.
   
Actually, collisions occur at Layer 2, not Layer 3, and the layer
  2
hardware automatically resends packets involved in a collision -
  layer 3
is never aware of it happening (although it may cause additional
  delay).
Eventually the ethernet card will give up if too many collisions
  occur
during retries, but this is very rare in practice unless the
  network is
*VERY* loaded.
   
 Assuming alaw/ulaw codecs in use (about 80k bps), a half duplex
  10 meg
 ethernet would handle roughly 20-25 rtp sessions before bumping
  into
   
the
   
 problem (your milage may vary). The majority of the folks on
  this list
 seem to be running home/soho systems and would likely never run
  into
   
the
   
 issue. But the heavier users will.
   
For a duplex mismatch, my experience is that if one end on a 100
  Mb/sec
link is half and the other is full, bandwidth is limited to about
  8
Mb/sec
max.  This is based on some tests I've accidentally conducted. 
  If you
try
to send 9 Mb/sec over that link, yes, some packets will get
  dropped as
they simply won't fit.  (But I do agree that for a half-half
  link, you
can
get about 20 Mb/sec)
   
--
Joel
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   ---End of Original Message-
  
  
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 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK
 
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Re: [Asterisk-Users] Cisco 7960 phones.

  What firmware did you upgrade to?
 
  If its version 5.0 and above the default password is cisco and to
  unlock it you press settings then 9.
 
  NO cisco's docs are simple.. You are just trying too hard.
 
 I want to thank you for the password of cisco.  It worked. I have
 finally gotten the phone to work.  Now to start setting all the new
 bells and tones it has!  I still think that they have over done the
 settings on the phone!

For those of us that have been around the block (at least a couple of
times), the quality of the Cisco product (produced by some other company
that I can't seem to remember ;) ) is significantly greater then a 
number of other devices, once you understand some of the values that
aren't noticed in 10 seconds or less.

Enjoy (... I don't sell/work for/sponsor Cisco products)

Rich


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[Asterisk-Users] CT1 and callerid

I'm just double checking.. I was told it wasn't possible but i'm going to
ask just in case.

Can you set outbound callerid on a channelized T1?

bkw
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Re: [Asterisk-Users] Merry Christmas, all Asterisk users!

 It's the day before Christmas here in Sweden, actually the night before 
 at this time...
snip 
 It's been fun spending the fall with the Asterisk project. I look forward
 to next year, with the new handbook coming in place, with many new 
 applications and features and - hopefully - many new Asterisk installs at 
 customer sites.
snip
 Have a wonderful Christmas, all of you!

And from all of us that have been around this list for a while, we
Thank YOU for taking the time and effort placed towards advancing the 
documentation and participation. You are the man!

May the almighty one grant speed to the wiki! :) (Absolutely no offence
intended; from one US swed to another swed!)

Rich


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Re: [Asterisk-Users] CT1 and callerid

2003-12-23 Thread Steven Critchfield

On Tue, 2003-12-23 at 19:22, Brian West wrote:
 I'm just double checking.. I was told it wasn't possible but i'm going to
 ask just in case.
 
 Can you set outbound callerid on a channelized T1?

I think there is a way to do something like DID with the 4 digits of
DTMF passed before the call. It is unlikely though that you will find
someone interested in doing that though. It is easier/cheaper to drop a
PRI into somewhere and then outbound caller ID isn't kludgey with DTMF. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

 7914's don't work with SIP.  SCCP only.  And why do people keep talking
 about this 8 dollar CCO account ... Its a service contract on the Cisco
 ATA-186.  The one for the 79XX's are over 80.00/yr

Careful Brian... things aren't always what they seem. There is some
flexibility built into their P/L plan! 



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Re: [Asterisk-Users] CT1 and callerid

HAHA you apparenlty aren't where we are... PRI is over priced... 3600/mth
SBC Victim... they have to backhaul it 110 miles.. where CT1's can be
served by the local CO.

bkw

On Tue, 23 Dec 2003, Steven Critchfield wrote:

 On Tue, 2003-12-23 at 19:22, Brian West wrote:
  I'm just double checking.. I was told it wasn't possible but i'm going to
  ask just in case.
 
  Can you set outbound callerid on a channelized T1?

 I think there is a way to do something like DID with the 4 digits of
 DTMF passed before the call. It is unlikely though that you will find
 someone interested in doing that though. It is easier/cheaper to drop a
 PRI into somewhere and then outbound caller ID isn't kludgey with DTMF.
 --
 Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Cisco 7960 Sounds patchy.

 I have gotten the Cisco 7960 working with my Asterisk system under SIP.
 The version is 5.03 that I am using.  Cisco Support said I should not
 upgrade to version 6 yet. My next question is the sound is patchy when
 people here me.  But I can hear them just fine not patchy.  I have the
 188 page Admin manual and it seem not to say anything about improving
 the sound. All other phones like IPDialog work fine without the patchy
 sound.   I have tried ulaw and alaw as the codex.  Both sound the same!
 Is there any other settings that can be done.  I remember that the
 X-lite has a transmit silence but I could not find this setting in there
 documentation.

Patchy sound has absolutely nothing to do with the 7960 software. All
versions from 2.1 through 6.0 have been solid as a rock from a usability
standpoint. If you have patchy sound, look towards improper configuration
of the phone vs asterisk definitions.

In my (somewhat biased) opinion, the 7960 is the top of the line once
you understand sip/rtp basics and provide the network infrastructure
to support the basics. (It's also the most expensive even for refurb
7960's. Your milage may vary.)

I avoided the Cisco v5 code for some time due to the back-level warnings
published by Cisco. However, I upgraded to v6.0 code and would recommend
it at a heart beat. It's been absolutely solid, stable, and they finally 
added some rather useful user features.

Rich


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Re: [Asterisk-Users] Cisco 7914 Expansion Unit (for 7960G IP Phone) Help With 7960's Speed-dials

The fun part is getting a clueful reseller on the phone to sell you the
correct thing.

bkw

On Tue, 23 Dec 2003, Rich Adamson wrote:

  7914's don't work with SIP.  SCCP only.  And why do people keep talking
  about this 8 dollar CCO account ... Its a service contract on the Cisco
  ATA-186.  The one for the 79XX's are over 80.00/yr

 Careful Brian... things aren't always what they seem. There is some
 flexibility built into their P/L plan!



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[Asterisk-Users] Outdialing with Voicetronix

2003-12-23 Thread Ahmad Faiz

Hi all,

Just thought I'd pass along some pointers when outdialing with Voicetronix's
OpenLine4 card.

I was having a tough time dialing out from *, it probably has something to
do with chan_vpb.c not waiting to hear the dialtone before telling the card
to dial. A quick fix was to insert a , in the dialstring telling the card
to pause before dialing.

However when the , was used in the dialstring, the Dial application
interpreted this as a command separator and was screwing things up. What I
did was to define

OUTDIAL=vpb/1-1/,,55

and used

exten = 555,s,1,Dial(${OUTDIAL})

to place the call. Works like a charm. Hopefully this bit of info may help
other VPB users out there.

Cheers,
Faiz


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[Asterisk-Users] Merry Christmas!

2003-12-23 Thread Michael Welter

Merry Christmas from the Colorado Organization for Victims' Assistance.

Our (Comdial) PBX fried after a power failure.  Thanks to Mark Spencer, 
Digium, VCCH, and the friends who support this group, we are now back 
on the air.

We wish everyone good health for the coming year.

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Re: [Asterisk-Users] Merry Christmas, all Asterisk users!

2003-12-23 Thread Miguel Cavazos

merry xmas olle and you all in the list happy holidays!!!

Miguel
On Tue, 2003-12-23 at 23:04, Olle E. Johansson wrote:
 It's the day before Christmas here in Sweden, actually the night before at this 
 time...
 
 We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
 merry-christmas-mode with the yet undocumented CLI command frosty-mode on, a mode
 where the PBX will connect all incoming extensions to the ho-ho-ho sound file and 
 then randomly
 pick a number in the +1234 country code (for the North Pole), dial out and bridge. 
 And these
 magical SIP connections will work over ANY type of NAT.
 (Due to the SIP header Santa-magic-cookie: on)
 
 And yes, the frosty mode is even un-documented on the http://www.voip-info.org wiki. 
 :-)
 
 It's been fun spending the fall with the Asterisk project. I look forward to next 
 year,
 with the new handbook coming in place, with many new applications
 and features and - hopefully - many new Asterisk installs at customer sites.
 
 It's snowing outside, the trees are already covered with snow and the stars are 
 glittering on
 a dark sky. My kids are sleeping, dreaming of their christmas gifts tomorrow. It's 
 going
 to be a traditional Swedish christmas...
 
 Have a wonderful Christmas, all of you!
 
 Warm regards,
 /Olle
 
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Re: [Asterisk-Users] MSN messenger and *

2003-12-23 Thread Balaji NJL

u can ignore the passport request.

u need to change the registry settings to make a phone
call. Do a search and
u ll find the details.
-B
- Original Message - 
From: Glen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 3:57 PM
Subject: Re: [Asterisk-Users] MSN messenger and *

 Speaking of MSN/Windows Messenger, how does one call
someone?
 Using the configuration specified, I've registered
it with Asterisk, but
 it requires that I add a Passport contact.

 Does anyone have experience calling a sip endpoint
without it being a
 Passport account?

 -g

 On Mon, 2003-12-22 at 20:42, Balaji NJL wrote:
  use this

  [3001]

  type=friend

  ;username=3001

  ;fromuser=Craig1

  ;secret=secret

  host=dynamic

  mailbox=3001

  context=sip

  dtmfmode=info

  auth=plaintext

  make sure ur MSN version is 4.7.0105.

  -B

  - Original Message - 
  From: Craig Waddington
  To: [EMAIL PROTECTED]
  Sent: Monday, December 22, 2003 10:10 AM
  Subject: [Asterisk-Users] MSN messenger
and *

  Sorry for the late reply.

  I try port 5060 and it just knocks me back
straight away, I
  cant see it even try to authenticate in
the CLI.

  X-lite works both inside the LAN and
outside using SIP.

  Messenger version = 4.7

  John I will try your suggestion with
sip.conf thanks for the
  help. I notice a few differences, I seem
to be missing some
  bits..

  Its like it is trying to authenticate with
the Linux box and
  not asterisk.

  Sip.conf

  [general]

  port=5060   ; Port to
bind to

  bindaddr=0.0.0.0; Address
to bind to

  context=sip ; Default
for incoming calls

  allow=ulaw

  allow=alaw

  allow=gsm

  allow=ilbc

  [3001]

  type=friend

  username=3001

  fromuser=Craig1

  secret=secret

  host=dynamic

  mailbox=3001

  context=sip

  dtmfmode=info

  I found 3 guides and each one seems to be
a bit different and
  use different ports.

  I am using the X100P, it is a home system,
to reduce call
  charges for my family overseas.

  If  I can get Messengger working it will
be easier to talk
  them through the setup.

__
  Do you Yahoo!?
  Yahoo! Photos - Get your photo on the big screen
in Times Square

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Re: [Asterisk-Users] MSN to GS - Call drops in 10 secs

2003-12-23 Thread Balaji NJL




i tried with only GSM too. With only GSM it doesnt 
even connect to GS. Then someone recommended to use ulaw and alaw and that 
helped. But the call drops after 10 secs. i did a 'sip debug' and what i found 
is that MSN doesnt even recognize that call is in progress 
and then drops the call. Any way i can increase this or disable this 
option.

thanks,
-B

  - Original Message - 
  From: 
  Craig 
  Waddington 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, December 23, 2003 4:34 
  PM
  Subject: RE: [Asterisk-Users] MSN to GS - 
  Call drops in 10 secs
  
  
  Balaji,
  
  I also have 
  the same issue. Works fine on any phone except GS for 
  me.
  
  After a bit 
  of research I found a post saying set the phone to offer only one codec 
  set.
  
  It looks 
  like we have to set the phone to use one codec  GSM 
  
  
  I am 
  concerned that you cant use passwords when logging in to * using 
  Messenger.
  
  Craig.
  
  
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Balaji NJLSent: 23 December 2003 23:04To: 
  [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] MSN to GS - 
  Call drops in 10 secs
  
  
  resending.
  
  
  
  Can anyone help me in trying to 
  understand what would be the problem. appreciate ur time. i need to get 
  this working.
  
  
  
  thanks a 
  lot,
  
  -B
  

- Original Message - 


From: Balaji NJL 


To: [EMAIL PROTECTED] 


Sent: Monday, 
December 22, 2003 8:15 PM

Subject: 
[Asterisk-Users] MSN to GS - Call drops in 10 
secs



Hi 
All,



i dont know what changes i made 
recently but i am unable to hold the call for more 10 secs between MSN and 
GS. PSTN to MSN and PSTN to GS and vice versa works fine. i am not behind 
NAT.Also MSN to MSN works fine too.



my SIP 
details



[general]port = 
5060bindaddr = 0.0.0.0context = bogon-calls;context = 
defaultdisallow=allallow=ulawallow=alawallow=ilbcallow=gsm



;My SIP phone - 
GS[2000]type=friendusername=2000secret=qweqwehost=dynamiccontext=from-sipmailbox=2000dtmfmode=inband



;MSN 
Msgr[2002]type=friendhost=dynamicinsecure=yesdtmfmode=inband;dtmfmode=rfc2833context=from-sipmailbox=2002;auth=plaintext

i did a SIP 
trace



it says 
Format=UKN

CSeq=BYE



thanks for the 
help,

-Balaji



Do you Yahoo!?Yahoo! Photos - Get 
your photo on the big screen in Times 
  Square
  
  
  Do you Yahoo!?Yahoo! Photos - Get 
  your photo on the big screen in Times Square

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Yahoo! Photos - Get your photo on the big screen in Times Square

Re: [Asterisk-Users] Merry Christmas, all Asterisk users!

2003-12-23 Thread Richard Lyman

this is a time to reflect, and i have much to reflect for come the end 
of a year.

for all those that i've pissed-off through out the year with nasty 
comments and such...

merry christmas to all.

sorry  (but come the 1st it's a new year and therefore can create a new 
list to atone for) G

Miguel Cavazos wrote:

merry xmas olle and you all in the list happy holidays!!!

Miguel
On Tue, 2003-12-23 at 23:04, Olle E. Johansson wrote:
 

It's the day before Christmas here in Sweden, actually the night before at this time...

We celebrate Xmas on the 24th, so I'm about to log off and switch my Asterisk into
merry-christmas-mode with the yet undocumented CLI command frosty-mode on, a mode
where the PBX will connect all incoming extensions to the ho-ho-ho sound file and 
then randomly
pick a number in the +1234 country code (for the North Pole), dial out and bridge. And 
these
magical SIP connections will work over ANY type of NAT.
(Due to the SIP header Santa-magic-cookie: on)
And yes, the frosty mode is even un-documented on the http://www.voip-info.org wiki. :-)

It's been fun spending the fall with the Asterisk project. I look forward to next year,
with the new handbook coming in place, with many new applications
and features and - hopefully - many new Asterisk installs at customer sites.
It's snowing outside, the trees are already covered with snow and the stars are 
glittering on
a dark sky. My kids are sleeping, dreaming of their christmas gifts tomorrow. It's 
going
to be a traditional Swedish christmas...
Have a wonderful Christmas, all of you!

Warm regards,
/Olle
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[Asterisk-Users] DTMF A,B,C and D