[Asterisk-Users] Can somebody with HEAD please test MOH on agent calls? M3976

[Nick Bachmann]

See http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0003976.
Essentially, the bug is that if a callback agent puts a caller on hold, 
that caller does not hear MOH.  This bug has been around for a while, 
but nobody has been able to follow through on testing to the point where 
we could nail it down.  If somebody with HEAD could verify that the 
problem does/doesn't exist in head, that would be much appreciated.

Please post your findings in Mantis.
Thanks!
Nick
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Re: [Asterisk-Users] Survey: what's the best HTTPd/TFTPd/FTPd to serve up configuration files to sets

[Nick Bachmann]

Jim Van Meggelen wrote:
I would like to start a discussion centred around the various ways one
might serve up configuration files from an Asterisk server (I know, it's
 

[snippage]
I have heard that khttpd is pretty lightweight, but its use seems to
have been deprecated, and it does not appear to be actively maintained.
Is TuX the way to go?
 

TUX, unless it has changed since I last looked (and I don't think it 
has) has to hand off requests for dynamic content to another 
full-featured server, like Apache. If you want to use any PHP, mod_perl, 
or CGI scripts (vmail.cgi for instance) on your web server, then you'll 
need two web server. 

As for tftpd and ftpd, I'm just not sure. Leightweight is the key, here.
The answer to this question is the same as the answer to which 
distribution is best: "The best one to use is the one you know."

Personally, I try to use Postfix, Perl, Apache, and PostgreSQL on 
Linux.  There are advantages to qmail, PHP, TUX, and MySQL, but for 
various reasons, I have chosen to get to know Postfix, Perl, Apache, and 
PostgreSQL.  Because I know them, I can set them up to perform more 
efficiently and more securely than I could replacements for them that 
are marketed as "more secure" or "more robust."

Tftp (and to some extent ftp) servers are sort of an exception to this 
rule.  There's so little to configure that I'm fine using whatever the 
stock is with my distribution.  So long as they're not totally 
bone-headed, they'll all do the job about the same.
  
Nick
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Re: [Asterisk-Users] Polycom Phone Calling Party ID

[Nick Bachmann]

Mark Floyd wrote:
I am trying to get the name and number to show up for an incoming calls on
my Polycom IP 500.  Right now only the name shows up, but in the call list
both name and number show up.  Any help on what to change in the config file
would be greatly appreciated.
 

Watch the display.  Once you answer the phone, the number should show up 
right below the name.

Nick
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Re: [Asterisk-Users] Can agents login be permanent across Asterisk restarts ?

[Nick Bachmann]

Robert Rozman wrote:
Hi,
I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is
restarted. Can this be avoided in some way ?
 

As Kevin insinuated, there is support for this in CVS Head.  It's called 
persistentagents and is set through agents.conf:

; Define whether callbacklogins should be stored in astdb for
; persistence. Persistent logins will be reloaded after
; Asterisk restarts.
;
; persistentagents=yes
The default it off, so you'll have to uncomment it to turn it on.  There 
was also a patch contributed in the bug report (M3202) for CVS stable 
that may still apply, if you're not running HEAD and don't want to upgrade.

Nick
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Re: [Asterisk-Users] Encrypted VOIP?

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
If I remeber correctly, Mark Spencer is working on encryption in IAX2
 

Sort of.  Some IAX encryption code went into CVS a while back, but it 
was more of a "talking point" than anything else, meant to give 
interested developers a starting point.  The -dev and -security list 
archives have some discussion about this.  I don't think there's been 
too much continued work on it, but volunteers are always accepted!

For SIP, native encryption should be done through SRTP, which many 
people have asked for but nobody has really delivered.  Again, you will 
find some good background discussion in the list archives.

Running IAX over stunnel would probably be feasible if both sides of a 
tunnel were machines. That's at least a little closer to to native, and 
very easy to set up.  However, I don't think that's what you were 
looking for... :)

Nick
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Re: [Asterisk-Users] How to download CVS with attended transfers

[Nick Bachmann]

Mike Sander wrote:
Hi
I know that attended transfers are only available in the CVS Head.
 

The version info reports:
Asterisk CVS-v1-0-02/03/05-10:24:22
 

You don't have HEAD.  Follow the instructions on asterisk.org to 
download from CVS HEAD (as in, not -r v1-0).

Nick
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Re: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCardApplicationforAsterisk

[Nick Bachmann]

Bruce Komito wrote:
 That's your opinion, and I'm sure you have good reason for it.
 However, in order to be widely accepted, any app must support mysql,
 simply because many environments run mysql as their choice of
 database, and are not likely to change.
Ah yes, the "pack mentality" logic for software.
I'm trying to think what other situation I've heard this used in...
Nick
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Re: [Asterisk-Users] UPS for Asterisk

[Nick Bachmann]

Andrew Kohlsmith wrote:
 On January 23, 2005 04:04 pm, Mike Sander wrote:
> Is the harddisk activity on a standard asterisk install such that I
> don't really have to worry if the power cuts??
 Not typically; there isn't much writing going on, this is true. Are
 you that cash strapped that a $75 UPS with a serial port is out of
 your budget?
No kidding... the cost of a server than won't come up again is much more 
substantial than the countermeasure... the $75 (you can get a 350 Va for 
$45 even!) and a slightly less energy efficient system. If you can 
afford to spend more, a decent active UPS would keep your power 
conditioned as well...

> As I understand, if HD activity is minimal, the probability of HD
> failure is significantly reduced.
 HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD 
manufacturers have done "head unloading" and such recently, but the risk 
is still higher if power is suddenly lost during a write.

Nick
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Re: [Asterisk-Users] Anyone know where a good source of mailing list stats might be found?

[Nick Bachmann]

Kanwar Ranbir Sandhu wrote:
On Sat, 2005-22-01 at 12:19 -0500, Jim Van Meggelen wrote:
 

I'm curious to know how the volume of Asterisk-Users rates as far
mailing lists go. This list sees over 200 messages per day, which has
GOT to put it in the top 5%, doesn't it? I'd love to know if anyone has
knowledge of any organization that might maintain such stats.
   

Someone on the Fedora mailing list wrote a script (or scripts) to
collect stats on that mailing list.  It compiles data on the top
posters, the mailer's used and other cool stats.  Every month the
results are posted to the list.
Either run a search or post a message there to get a hold of the script.
It should be very easy to adapt it to this list.
 

There is such a thing that gets send weekly in comp.unix.admin that's 
from script archive... I point this out because the name of the script 
archive is in the email, so if somebody wants to do this, they don't 
have to look far. :)

NB
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Re: [Asterisk-Users] Re: zaptel on 2.6.10 kernel - debian.

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
Hi Guys, Gals.
Ok, so I have latest CVS sources on a debian box, 2.6.10-1-386 kernel
kernel headers isntalled in the right plauce and all that stuff .. but 
whatever I try .. same results, I only need to get ztdummy working for 
a conference .. but I always end up stuffed :(
heres the compile:
Call off the hounds, I think the problem is fixed in M3391...
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[Asterisk-Users] Come join the Asterisk Bookclub

[Nick Bachmann]

Greetings all-
For whatever reason of personal insanity, I've decided to start an 
Asterisk bookclub.  Basically, we'll pick three books every month (a 
users book, a developers book, and another general interest book) and 
then read and discuss on IRC in the #asterisk-bookclub channel. 

The users' book will be something related to telephony or the technology 
related to being an Asterisk administrator.  The developers' book will 
be, similarly, related to software development with a particular focus 
on telephony, drivers, and the related technology an Asterisk developer 
should know about.  The general interest book will be something fiction 
or nonfiction that is interesting to the folks who hang around here :-).

To get things rolling, here are our books for January:
Users: _Ethernet: The Definitive Guide_, from O'Reilly.  ISBN 1565926609
Developers: _Secure Programming Cookbook for C and C++_, also from ORA. 
ISBN 0596003943**
General Interest: _Cuckoo's Egg_, by Cliff Stoll. ISBN 0743411463

So, head out to your favorite bookstore and join #asterisk-bookclub!
See ya on IRC,
Nick
(IRC hermie)
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Re: [Asterisk-Users] Realtime

[Nick Bachmann]

Serge Schumacher wrote:
Hi,
 

Jan  6 01:43:09 WARNING[12209]: pbx.c:796 pbx_find_extension: No such 
switch 'Realtime'

 

 

What does this message mean ?
 

Something wrong with the switch statement in my extensions.conf or 
maybe is the module net correctly installed ?

Perhaps you might consider posting your extensions.conf and other 
relevent troubleshooting details?
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Re: [Asterisk-Users] final call for Departments

[Nick Bachmann]

Alspach Family wrote:
Today is the day.  The most up to date list is attached. I will 
forward it to Rob Friday night so, anything you want and can get to be 
by then I will add.
Do the phrases being send include those in bug 3006?  If not, can they?
Lets say half of them are active readers and half of those are using 
Asterisk seriously.  If those 250 people remaining could just donate 
$.50 each, we would reach our goal..
Public television and radio stations everywhere would be proud of you.
Nick
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Re: [Asterisk-Users] WARNING[22314]: No such switch 'Realtime'

[Nick Bachmann]

Gabriel Afana wrote:
Ahhh, and I've read every message telling everybody they dont have the
lastest version...thats why I went to asterisk.org and downloaded the
highest-number version I could find in the FTP Okdownloaded latest
CVS but now asterisk wont compile.  I had it working before.
during "make", it says:
chan_zap.c:61:2: #error "You need newer libpri"
This error means that you need a newer libpri.
That means you need to:
# cvs checkout zaptel libpri asterisk 

and then
# cd zaptel
# make clean; make install
# cd ../libpri
# make clean; make install
# cd ../asterisk
# make clean; make install
Nick
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Re: [Asterisk-Users] g711 ulaw vs alaw

[Nick Bachmann]

Tracy R Reed wrote:
On Thu, Dec 16, 2004 at 01:11:34PM +0100, Roy Sigurd Karlsbakk spake thusly:
 

I think there is a bit more difference. The byte code of ulaw is a
monotonic function of the amplitude whereas in alaw the code is xor:ed
with a bit mask of 0x55.
 

Wow! Encryption!
   

Scary thing is, it would be illegal under DMCA to un-XOR it! (In the US anyway)
 

Amusing, but offtopic, incorrect, and misleading...
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Re: [Asterisk-Users] 191st simultaneous call fails

[Nick Bachmann]

Nick Bachmann wrote:
Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls.  This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call fails every
time with errors like:
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1
Dec 14 15:44:00 WARNING[1215]: Failed to create update thread!
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, 
span 9
Dec 14 15:44:00 WARNING[1215]: Call specified, but not found?
Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9

It's not tied to which channel the call comes in on.  It's some
resource that's exhausted after 190 calls.  A limit on threads?
 

Try http://people.redhat.com/alikins/tuning_utils/thread-limit.c and 
see what happens.
Also, have a look at the PPT on http://www.astertest.com/  They included 
some tweaks they used to increase call volume.

Good luck, and please report your findings back to the ML & Wiki!
Nick
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Re: [Asterisk-Users] 191st simultaneous call fails

[Nick Bachmann]

Jim Gottlieb wrote:
I've been testing both T400P and TE405P boards and I'm running into
some kind of hard limit on the number of simultaneous calls.  This is
on x86 with 2 Athlon MP 2800+ CPUs running Fedora Core 1.
Everything is fine up to 190 channels, but the 191st call fails every
time with errors like:
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1
Dec 14 15:44:00 WARNING[1215]: Failed to create update thread!
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9, span 9
Dec 14 15:44:00 WARNING[1215]: Call specified, but not found?
Dec 14 15:44:00 WARNING[1215]: Hangup on bad channel 0/9 on span 9
It's not tied to which channel the call comes in on.  It's some
resource that's exhausted after 190 calls.  A limit on threads?
 

Try http://people.redhat.com/alikins/tuning_utils/thread-limit.c and see 
what happens.

Nick
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Re: [Asterisk-Users] My Boss wants background music!!!!

[Nick Bachmann]

Satchid wrote:
Dear Members,
I am searching for a new PBX for the company. My choice is Astrisk. My Boss
wants background music via all the telephones. This is done in a
conventional PBX that he wants, but I can use the Asterisk PBX if it can do
this also. 
As I said he needs background music on every telephone this is not to be
mistaken with music on hold. 
 

How about modifying the chan_agent stuff?  Right now, if an agent logs 
into a queue, he hears music until a call comes into him.  So you have 
the option of making a queue for every phone (which wouldn't be all that 
great) or creating a new application that copies some of the agentlogin 
functionality, perhaps hanging up when a new call comes in?

The bit stream is an MP3 file of 8 Kbs. At the server it might be at the
maximum 570Kbs if it has to send it individually to each telephone. 
The network: 10/100 layer-3 switches with QoS on a 1000Mbs backbone.

But the music isn't sent as an MP3 to the phone, it's sent using 
whatever codec you're using.

Nick
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Re: [Asterisk-Users] Ripping CD audio for MOH

[Nick Bachmann]

Matt Riddell wrote:
Thomas Johnson wrote:
Hello-
I've got some audio CDs that I'd like to use for MOH.

I just thought I'd point this out seeing as no one else has.
It is illegal to use an artist's music for music on hold without the 
copyright holders permission.

Since you brought it up, I'd like to point out that for a little less 
than $200/yr, most American small businesses can get a license from BMI 
for music-on-hold.  A BMI license covers their 4.5 million works, which 
tends to be pretty good in terms of diverse genre and age.

Nick
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Re: [Asterisk-Users] No ring signal when calling internal extensions ?

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
 can you make asteirsk do a fast ring as well?
Do you mean a fast busy?  If so, what you're looking for is called 
congestion.

If you mean making a faster ring cadence (time between rings), look at 
the indications.conf wiki page.

Nick
P.S. Please don't post that signature to the mailing list, especially if 
you're going to top-post.

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Re: [Asterisk-Users] ftmp header

[Nick Bachmann]

Matt Schulte wrote:
All,
We are using a SIP provider that is expecting 0-15 response for
fmtp. Our CVS Head asterisk server is sending 0-16, I looked up an rfc
and it stated:
RTP Payloads for Telephone Signal Events
   RFC 2833
   Henning Schulzrinne, Scott Petrack.
   May 2000
   Implementation notes:
   * Implementations can support events 0 through 15 (DTMF) by
simply ignoring the packets, but MUST declare all event numbers that are
meaningful to it in the fmtp parameter, including 0 through 15
 

How about opening a bug? 

Here's what I see in the source:
/* Indicate we support DTMF...  Not sure about 16, but MSN supports it 
so dang it, we will too... */
 snprintf(costr, sizeof costr, 
"a=fmtp:%d 0-16\r\n",

Nick
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Re: [Asterisk-Users] Ethernet Channel Bank idea

[Nick Bachmann]

nik martin wrote:
Anyone ever thought about an Ethernet based channel bank?  Basically a 
rack mount set of 24 IAXys?  That would be cool, IMO.  No wrangling 
with  zaptel, etc.  IAX as the * <-> Channel bank protocol.

These exist.  The Mediatrix 1124 is just one example.
Nick
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Re: [Asterisk-Users] Is Gigabit Ethernet necessary?

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
 For an office that is using VoIP phones to connect to Asterisk, is
 gigabit ethernet really necessary for the Asterisk box to connect to
 the switch? I know that I won't even approach the limits of 100 Mbps,
 but would gigabit help with latency / collisions when several calls
 are underway? The fact is, anything going outside the office will be
 over a data T1, so intuition tells me that 100 Mbps should be fine...
 The office will have 20 phones, with remote VoIP phones added to the
 mix later on.
http://www.voip-calculator.com/calculator/lipb/
Don't forget that you can't send 100 Mbps through a 100Mbps link.
 TIA,
TR41, probably.
Nick
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Re: [Asterisk-Users] FOP Asterisk Manager Login Failed?

[Nick Bachmann]

Noah Miller wrote:
 I've told lots of people about the Flash Operator Panel before, but
 I've never actually used it myself. I've got it all set up nicely,
 but I can't seem to authenticate to the asterisk manager (which is
 running on the same box). When I set the op_server.pl to give debug
 messages, it shows that it can reach the asterisk manager, but cannot
 authenticate:
 ** Asterisk event received, process block... -> Action: Login ->
 Username: user -> AuthType: MD5 -> Key:
 0be2f6f6e39f05a53f5a292517ede3e2
 ** End of block <- Response: Error <- Message: Authentication failed
 I note that it says the authentication is done with MD5, do I need to
 put an MD5 hash in for the secret in the configuration files?
No. The md5 is used so that your actual secret does not have to be 
transmitted in plaintext.  The concatination of the random key and the 
secret is computed by both sides and hashed, if these two intermediate 
forms of your secret are the same you are authenticated.

 [user] secret = usersecret deny=0.0.0.0/0.0.0.0
 permit=127.0.0.1/255.255.255.0
Is your FOP on a different machine?  If so, you'll have to explicitly 
add its IP or remove the deny statement, as it is blocking all IPs on 
all subnets.

Nick
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Re: [Asterisk-Users] Asterisk crashes my router!?

[Nick Bachmann]

Sean Cook wrote:
Sounds like it is time for a different router... There are a few routers
out there with buggy nat engines... they are fine when you are doing
typical nat, but if you are trying to do 1:1 nat... get a good router or
 

I've been very happy with Netopia 3386-ENTs.
make a BSD box to use as a router.  I would highly recommend
http://m0n0.ch/wall for a great do it yourself router
This was so funny that I had to share it:
"m0n0wall is probably *the first UNIX system that has its boot-time 
configuration done with PHP*, rather than the usual shell scripts, and 
that has *the entire system configuration stored in XML format*."

There's an excellent reason they're the first: those are both such 
unbelieveably terrible ideas, especially the PHP init scripts. 

I would reccomend IPCop, because their designers are a little more 
grounded.

Nick
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Re: [Asterisk-Users] Phone Selection

[Nick Bachmann]

Rich Adamson wrote:
 I'd certainly agree on the IP500. Got one sitting in front of me
 right now. Works great, but learning curve on configurations is
 greater then the Cisco's partially because all config parameters are
 in xml format
I agree, the configuration files are hard to learn, but they are very 
powerful if mastered.

 (and I can only edit with vi), and trying to find parameters that
 accomplish a needed function is a little difficult.
Having passed the Polycom certification tests for Soundpoint IP (that's 
a little joke to anybody who's taken them :-)), I can tell you that 
Polycom recommends a free program called XML Notepad from Microsoft to 
edit the files.

 Also had problems with a brand new phone; couldn't download software
 directly (only supported through authorized resellers). Polycom
 overnighted last Tuesday; haven't seen it yet. Support is almost
 patterned after Cisco.
My experience has been that their first level support and RMA process 
are adequate.  Not much more, not much less.  But the phones themselves 
are very solid pieces of equipment and very reliable.  I'd take them 
over Cisco any day, since they're so much cheaper for the same or 
similar hardware.  I've been especially happy with the speakerphone.  I 
also like them because they look like business phones, not something you 
bought for $10 at Radio Shack (*cough* Grandstream *cough*).

Nick
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Re: [Asterisk-Users] PRI Dialing failure?

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
*Smack*, you're right, changing the g3 to g2 help nicely.
But now the PRI seems to be refusing the call (Channel 0/1 got hangup):
--snip--
  -- Executing Answer("Zap/38-1", "") in new stack
   -- Accepting call from '' to '15123455476' on channel 0/14, span 2
Nov 28 16:08:14 DEBUG[1894]: chan_zap.c:1221 zt_enable_ec: No
echocancellation requested
   -- Executing Dial("Zap/38-1", "Zap/g2/15123455476") in new stack
   -- Called g2/15123455476
   -- Channel 0/1, span 3 got hangup
 

Try turning PRI debug on and looking at that... post back if you still 
can't figure it out.

Nick
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Re: [Asterisk-Users] Is "Busydetect" obsolete in the latest CVS?

[Nick Bachmann]

Garry Taylor wrote:
Hi Steven, 
 

Please don't top post, it breaks threading.
   -- Zap/2-1 is ringing
Nov 27 13:25:27 WARNING[6718]: chan_zap.c:3463 zt_handle_event: Didn't
finish Caller-ID spill.  Cancelling.
   -- Zap/2-1 is ringing
   -- Zap/2-1 is ringing
 

Open up chan_zap.c, change  DEFAULT_CIDRINGS to:
#define DEFAULT_CIDRINGS 2
Recompile & install.
I'll take Steven's $10 now :-)
And then I will have this PBX ready for my customer.
 


Nick
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Re: [Asterisk-Users] T1 and FX CPE

[Nick Bachmann]

Matthew Boehm wrote:
Could you give an example of a "cheap 1-port FXO gateway" ?  And yes, the
Cisco phones do have an "Emergency Proxy" and a "Backup Proxy" option.
 

(Please reply inline instead of top-posting so that threading is preserved)
I believe the SPA-3000s have an FXO port.  If you want a more solid 
answer, call one of the plentiful VoIP supply places (like Atacomm) and 
ask them what to get. 

A Google search on "FXO gateway" will get you far too.
Nick
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Re: [Asterisk-Users] T1 and FX CPE

[Nick Bachmann]

Matthew Boehm wrote:
Hey guys,
 Looking for some suggestions here on hardware to use.  We have several
business customers wanting to start using our VoIP service. We will be
replacing their 10 year old Panisonic system with Cisco 7940 phones and a
T1. Problem is, they have to keep 1 POTS line for directory listing and 911
emergency failover.
 I would like suggestions on how to accomplish this. Our asterisk cluster
is downtown so we can do the T1 between us and them easily. However, the
FX911 is the big problem. Lets say that the T1 line goes down and someone
tries to dial 911. Normally the call would go over the T1 down to asterisk
cluster and go out over PRI or an FX card. But if the T1 is down, you can't
do that. So, the POTS line needs to be plugged into something at the
customer site.
 

How about getting a cheap 1-port FXO gateway and setting that to be the 
secondary SIP gateway on all the phones?  I'm pretty sure you can do 
that with the Ciscos.

The other possibility, if their office isn't too large, is just to find 
a big red POTS phone and hook it up to the line.

There are a few Cisco routers that will accept both the T1 and the FX but
we can't intercept a SIP call at the router. Right now, the only solution we
can think of is putting an asterisk box at their location with an FX card.
That way if the T1 goes down, they can still dial intra-office and dial 911
out the FX card.
 

That's a valid option as well.
Nick
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Re: [Asterisk-Users] how to use stop calls

[Nick Bachmann]

michelle li wrote:
Hi:
I am new user of Asterisk. I can make phone calls. How
can I stop the call after it goes a certain time(eg,
20 seconds). I tried AbsoluteTimeout command, it does
not work. Can anyone help me? 
 

Can you post relevant sections of your dialplan... I suspect you do not 
have AbsoluteTimeout implemented correctly.  Also, you should always 
include what verison of Asterisk you are using.

Nick
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Re: [Asterisk-Users] Fw: TDMoE over bonded NIC's

[Nick Bachmann]

Kevin Brennan wrote:
Kevin Brennan wrote:
   

I am planning to configure * box A with PSTN interface to route faxes to
 

box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
 

2GB?!? Remember, each voice channel you trunk across TDMoE is 64Kbps.
While overprovisioning is laudable, I don't think box B can handle
15,000 circuits.
   

It happens servers come with twin GB NIC's, bonding is for redundancy not
capacity.
 

You know you shouldn't (can't?) use the same interface for regular IP 
networking and TDMoE, right?  The TDMoE should have an address-less NIC 
to itself and _really_ shouldn't run through a hub (an xover would be 
ideal).  Bonding seems possible, since the bonding is done at a lower 
level than Asterisk knows about and just creates a virtual device.  
Because both sides just see one NIC with one MAC, the only problem I 
could anticipate would be increased latency, but I doubt it's as 
significant as the latency a switch creates.

Nick
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Re: [Asterisk-Users] Fw: TDMoE over bonded NIC's

[Nick Bachmann]

Kevin Brennan wrote:
I am planning to configure * box A with PSTN interface to route faxes to *
box B (running spandsp) over TDMoE. I am using 2xGb bonded NIC's for
connection between servers.
2GB?!? Remember, each voice channel you trunk across TDMoE is 64Kbps.
While overprovisioning is laudable, I don't think box B can handle
15,000 circuits.
Nick
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Re: [Asterisk-Users] How to enter billing codes when dialling

[Nick Bachmann]

AHBLWEB wrote:
That's good and will get me the account#.
I guess it's too much to ask for the same kind of tone prompts that occur
after dialling the 77 (boop-boop) and then after the 7 digits
(beep-beep-beep) before the user gets the final real dial tone?
 

If you want another dial tone, set up some contexts and use DISA to give 
new dial tones. 

However, if you can retrain your users there are better ways to 
accomplish what you're after (i.e. just one DISA() with a password file) 
than emulating your old PBX exacatally.

Nick
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Re: [Asterisk-Users] A new alternative to see who is online

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
Hi all,
I have been facing about the problem to know who is online with asterisk PBX.
However users wanted to see it right away, without launching any application. 
As I could not find any solution with IP phones and users were really 
complaining, I decided to write this little application that runs under windows 
and stays on screen.
Nicolas -
Good work with Leon! 

Could I just make one feature request?  How about the ability to tuck 
the application into the system tray, so you could just click its icon 
and have a vertical display pop out.  Kind of like the GNOME Clock (see 
http://www.not-real.org/scr.html if you're not familiar)...

I think this would be handy for those users who don't like a cluttered 
desktop...

Nick
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Re: [Asterisk-Users] How to enter billing codes when dialling

[Nick Bachmann]

AHBLWEB wrote:
Our current ROLM switch uses two-digit Feature Access Codes (FACs) for 
long-distance calls to force the entering of a set of seven digits 
representing a client and matter number before giving a real dial 
tone.  This sequence is passed as part of the CDR record and is used 
to charge the phone call to the correct matter.
 
Looking through the Asterisk documentation I can see how a fixed 
billing code can be assigned to an extension so that the caller gets 
charged but how can I set up a dialling plan similar to the above 
without having to write a script to parse the matter number out of a 
stanard CDR record?
 
For example, a call to New York which should be charged to matter# 
1234567 is dialled as;  77-1234567-1-212-555-1212.

http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf
exten => _77XXX1NXXNXX, 1, SetAccount(${EXTEN:2:7})
exten => _77XXX1NXXNXX, 2,Dial(Zap/g1/${EXTEN:9:11})
With this, the account code will at least be a separate field in your 
CDR record... I think that's what you're asking for...

You can add some logic to check for valid account codes and all that.  
Have a look at the AstCC and the Authenticate application for some ideas.

Nick
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Re: [Asterisk-Users] SS7 for *

[Nick Bachmann]

Steve Underwood wrote:
Hi Angel,
It is working pretty well. I think it will be available about the end 
of the year. I will not be free. It will be supplied with a 
commercially licenced Asterisk.

Here's a question: if the author has purchased a commercial license to 
use Asterisk, and I get binary modules from him, I can still use them 
with my CVS-based Asterisk, right?

Nick
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Re: [Asterisk-Users] SysMaster and GPL Violation (lets think before we

[Nick Bachmann]

Benjamin on Asterisk Mailing Lists wrote:
However, I did not sign the disclaimers because it also asks for any
future rights. That's really stupid. If you were to invite me to a pub
and offer to buy me a pint of Guinness, then would it not be
unreasonable of me if I said, "I accept, but only on the condition
that you will pay for all beers in the future, too."
 

I believe that's why http://www.digium.com/disclaim.changes exists.  But 
then you'd have to disclaim every change every time... that's why it 
says "both documents are fine, just whichever you are more comfortable 
with" on the Bugs page.

At least, that's my understanding...
Nico
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Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System

[Nick Bachmann]

Henry Devito wrote:
<>I know I am top posting and that is a no no, but I would like to 
comment on
this generally. I just did this with a historic building with the same
situation cat3 two pair in each office. I used a Tut systems solution
called expresso this gave us cable TV and Ethernet to each office over the
existing cat 3. Amazing technology I think. I'm not affiliated with them
at all. I think their website is http://www.tutsystems.com

-Original Message-
[snip]
<>You do have other options, such as products like Tut's
(http://www.tutsys.com/mtu/products/expressomdu/index.cfm), 

It's funny that if you hadn't of top posted, you might have seen my link 
to Tut's Expresso product...

Nick
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Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System

[Nick Bachmann]

Michael Welter wrote:
We have a 100 year old building here in Colorado that needs a new 
telephone system. The building (five floors) is steel frame with lath 
and plaster walls. There is no crawl space above the ceilings or under 
the floors.  The building is "historic", and nothing can be done to 
the exterior.

The current system uses existing Cat3 (two pair) to get to the digital 
telephone set in each office.  Some offices have an additional pair 
which is used for fax (and DSL).  I belive this fax line is a POTS 
line from the telco.

The owners would like to replace the existing telephone system, but 
they are adamant that the exsiting wiring be reused.  

How about ADSI phones?  You can use the Cat 3 and still have fancy phones.
They would like to provide a LAN connection to each office for both 
data and voice.  (They would also like to install cable TV in each 
office, but cable install costs would be $80,000+.)

The owners are concerned about frequent power failures and keeping the 
telephones operational.  Whatever equipemnt and telephone sets we put 
in the offices will have to be powered from a central UPS (PoE).

Most ADSI phones will still allow basic telephone use (you can make 
calls but no display or lights) when the power goes out.

So how can I do this?  Can I use RS485 adapters to get ethernet to 
each office via the two pair?  What kind of data rate can I get with 
RS485, and would it be half- or full-duplex?  

This would be possible, but is the least desirable of any possible 
option, since you can't really hope for more than 10Mbps H-D, since 
you're dealing with fewer twists and, likely, inferior cable 
construction.  You may have better luck on longer runs, but remember, 
adequately cabled Cat-5 isn't designed to go over 100m.  Based on the 
old buildings that I've cabled, you rarely get a straight shot.

Would wireless work in a steel building? Is there some other 
technology that can be used?

Wireless would be a good choice, especially if the building has a steel 
skin.  Without ever seeing the building, my recommendation would be to 
put a good 3Com or similar (NOT LINKSYS) AP staggered through every 
floor (i.e. not directly above the lower floor's... shift for greater 
coverage area) connected to your wiring closet with fiber.  Since you'll 
only have 5 runs, this shouldn't cost too much.  Allied Telesyn* media 
converters will set you back =~$150/end, or you could (preferably) get 
fiber cards for a switch which cost about the same. 

If you still wanted to do a cable based networking, you could just run 
the fiber to a small switch on each floor and figure out how to 
discretely feed cable to each office, but I can almost guarantee the 
solution will come down to conduit, which is hard to make look good.  At 
least with wireless, you only have one cable per floor to add and you 
can figure out how to put your AP in a discrete place.

You do have other options, such as products like Tut's 
(http://www.tutsys.com/mtu/products/expressomdu/index.cfm), but I think 
you'll find them expensive and limited.

If you want more details on how I've done stuff like this, feel free to 
email me off-list.  I also do consulting work, if you're interested...

Nick
*I recommend this particular brand because I've used their converters in 
lots of applications (including 10+ mile building-to-building runs) 
without ever seeing one die.  And they're pretty cheap.
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Re: [Asterisk-Users] Hold music while ringing

[Nick Bachmann]

Matthew wrote:
What im after is a dial plan, so when a user calls into a 'specific' 
number, instead of hte meharing hte ringing until I pickup the call on 
my SIP phone.

Tried looking thru voip-info but the clsoest i could find was a 
WaitUserOnHold in teh dialplan, not sure if this is what im after nor 
how to implement it

Could you repost that as  coherent message?  I think what you're looking 
for is right in voip-info.org, but I really can't know until I 
understand what you're asking. 

Nick
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Re: [Asterisk-Users] DISA() anyone?

[Nick Bachmann]

Michael George wrote:
I'm having some trouble with DISA() in a call plan that worked before 1.0.  If
anyone has experience with it, I would appreciate some advice.
 

Perhaps you could post relavent sections of your dialplan...?
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Re: [Asterisk-Users] KSS/BLF on Asterisk

[Nick Bachmann]

[EMAIL PROTECTED] wrote:
Folks,
I am trying to determine the best way to allow a station to monitor the
status of another station.
For example: 
a reception set needing to see the status of 20 or 30 phones
OR
an executive assistant wanting to have appearances of several other
extensions, in order to monitor their status and assist with call
handling.

I know Snom has a phone that you can attach an add-on module to, but I
don't know how you'd program Asterisk to deliver status information to
those buttons.
 

Get a hint! :-)
Check out the "hint" priority in extensions.conf.  There are also some 
details in the wiki.

Nick
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Re: [Asterisk-Users] Medium volume 100% SIP/IAX PBX.

[Nick Bachmann]

Marconi Rivello <[EMAIL PROTECTED]> wrote:
> Hi, I have a curiosity: how much does a regular PBX system cost? I'm
> curious if using IP telephony in a building is cheaper than a regular
> PBX, because of the high cost of the IP phones.

Take a look at
http://www.buyerzone.com/telecom_equipment/phone_systems/buyers_guide7.html.
As the article pointed out, TCO is important as well.  Commercial PBXs
usually require technicians with special software and training that can
run >US$100/hr.  If you already have a sysadmin, the savings in labor
alone can pay for an * system over a few years.
The prices for cards and software modules can add up quick as well... even
if you could get a cheap NEC/Panasonic/Nortel/Avaya for the same cost as
an IP-PBX, you'd basically have a featureless switch with featureless
telephones.  At that point, you might as well call up the local telco and
get Centrex service.
Figure that very nice IP phone can be had for around $250, and nice * box
can be built with dual processors for about $3000 that can handle 30
users.  That equates to about $350/user for a somewhat over-provisioned
all-VoIP setup.  That's far less than buying a $900/user PBX.
Nick


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Re: [Asterisk-Users] Asterisk for a large scale implementation

[Nick Bachmann]

> Is there a good place to find Asterisk consultants?

There is an asterisk-biz mailing list.

Nick


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RE: [Asterisk-Users] E1 Red Alarm

[Nick Bachmann]

> Hi.  Just for grins, reverse the BNC's.  You could have the
> tx and rx crossed, happens all the time on ds-3's.  Also,
> with the BNC's unplugged and hanging, You should have a
> Yellow light on the Hdsl Box (I've always called that a
> smartjack, but we call football soccer, so what do we know?

Not much, the way some tell it :-)

The HDSL box gets a red light when the BNCs are unplugged or switched, or
when i try to mess with the cabling (i.e. use a straight-thru) between the
balun and the TE410P.  The one thing I haven't tried doing is seeing if
tip and ring are backward, but it seems a remote possibility.
> :) ).  If you don't get a yellow light with the BNC's
> hanging then the smartjack probably has an internal loopback
> set and you need to give your telco a call to clear it.

I thought about this, if a replacement balun and new coax don't fix the
problem, the guy over there said he'd try to get them to come out.  Much
like American public utilities, though, he said that the local telco
doesn't much like housecalls.  Hopefully the balun fixes it and we don't
have to deal with that eventuality.
Nick


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Re: [Asterisk-Users] Woodpeckers

[Nick Bachmann]

> Hi Michael,
>
> Michael Welter wrote:
>>  the codecs I use?  Filter-out everything between, say, 55 and 65Hz?
>
> Notching may not be that effective, as it will not deal with the
> harmonics. The analogue to digital converter should already be
> filtering  below 300Hz, so you probably have quite a lot of hum if it

300Hz is pretty high to filter out... it's still well within the rage of
voices. To compare, 300Hz is about a diatonic concert D.
> gets through  that. There isn't much you can do to eliminate the hum
> your callers  hear. For your own hum, additional filtering may be
> beneficial. This  could be implemented by analogue or digital filters.
> Simple RC analogue  filters probably would not be sharp enough to give
> much benefit, so you  would need a more complex active analogue filter.
> A digital filter would  require no hardware changes, but you need
> someone who knows what they  are doing to implement it for you.

Then how about something like http://www.digitimer.com/research/quest/? 
It seems to take care of harmonics and specifically targets the 50/60Hz
range.
Nick


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Re: [Asterisk-Users] Surveys

[Nick Bachmann]

>> Is it possible to have the system outdial and take surveys. either by
>> receiving DTMF or voice?
>
> Yup.  Just have the system use the outgoing queue (see sample.call) and
> have  it call an AGI script upon answering.

If you want CDR data, be sure you connect to an extension that starts the
AGI.  Directly connecting to an application, like agi, will result in no
CDR being collected.
Look at:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out

I used this as a guide when creating an application that did auto-dial out.

Nick


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Re: [Asterisk-Users] Send DTMF tone Like 'C' on connected call

[Nick Bachmann]

> Dear to all
>   someone know how is possible to have a DTMF tone like "C" AKA Alpha
>   Tone
> (connect tone) to the caller?

Yes, it's possible.


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Re: [Asterisk-Users] Digium X100P for $43

[Nick Bachmann]

> Digium X100P / new cards are is available on ebay for $43.
>
> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=3073050567&category=3309
> >
>
> Hope this helps to who want to play with X100P! Are these being sold by
> Digium ? I don't know ??
They're not Digium, they're compatible knock-offs.  The seller is a bit
duplicitous about that fact, but they're not genuine Digium as far as I
know.
People have reported that they work fine, but I still wouldn't trust them
for a true production system.
Nick



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Re: [Asterisk-Users] Lucent Definity + Asterisk Success!

[Nick Bachmann]

> Hey Everyone,
>
>   Thanks for helping with this integration, I finally got it working!
> I have inbound and outbound dialing from/to the asterisk box via a pri
> line! I would definitely like to contribute this information to the
> wiki so we can fill in some of the information gaps there. In the short
> term I wrote up what I did and scanned in all of the nescessary screen
> prints from the lucent into a pdf document. I will probably put it up
> on the web in the next few days, but in the meantime if anyone wants it
> just send me a message and I will forward it to you.

Would you be interested in contributing to the Asterisk book? We have a
section about PBX integration that's pretty empty.  We have a mailing list
([EMAIL PROTECTED]) an IRC channel (#asterisk-doc on
irc.freenode.net) and a website (www.asteriskdocs.org).  If you can get
your data to people through one of those channels, we can probably take it
and format it from there for you, if you don't have the time or interest.
Nick


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Re: [Asterisk-Users] I stumbled on this list...

[Nick Bachmann]

>
> Hi there,
>
> I stumbled on this list mostly by accident.  I came across Asterisk *
> as a means to help me get a better handle on my soaring telephone
> costs.  Each month I look at my phone bills and my stomach just turns
> because I can not find any competition to Verizon which is the local
> anointed phone company around here.

Trust me, it could be worse.  You could have Ameritech, for example.  Or
Quest. In fact, I'd say Verizon is probably the best of the major
RBOCs/ILECs (I'm in former GTE country, not NYNEX/Bell Atlantic, which
might be different).  But even then, they do leave something to be
desired.
> Since I am a neophyte at all this I was wondering if some kind soul
> would confirm/disconfirm my assumptions about this software called
> Asterisk *.
>
> 1)  Am I correct to assume that there is a way to dump Verizon and
> strictly go VOIP in a SOHO situation?

Yes.

> 2) Can 1-800 numbers terminate to a VOIP assigned number?

This a service of the provider that many do offer.

> 3) With VOIP am I under the assumption that one must also purchase
> licenses for such service to work.

No, you don't need to purchase licenses.  Asterisk is published under the
GNU General Public License.
> 4) Who are the companies I can purchase VOIP service from?  I need
> numbers in my local area code, plus I need some kind of unlimited VOIP
> service Asia - mainly to Taiwan.

Google can help with this. There are a couple of "Big" providers such as
NuPhone, Voicepulse, etc.
> 5) Am I being unrealistic in my savings by implementing an Asterisk *
> PBX in our SOHO situation.

There is not doubt, that VoIP can save you money.

If you're talking about a number or two, you'd be best to stick with just
getting an ATA (a little box that turns your phone line into VoIP) from
your provider, and not messing with a full fledged PBX.  But if you have a
PBX or Key System already, you might consider using Asterisk.
As a warning, however, if you don't know much about (pick two) telephony,
Linux, or software development, and you're still interested, you might to
well to find somebody (like, say, me :-) who can give you a turnkey
system.
If you do want to go it alone, there is a book at
http://www.asteriskdocs.org/, a wiki at http://www.voip-info.org, and a
search engine at http://search.voip-forum.com/.  Please utilize these, as
they will answer most of your questions about *.
Nick


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Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

[Nick Bachmann]

>>> Andrew Kohlsmith wrote:
>I would set the "Enterprise Class" bar at five 9's reliability
>(about 5.25 minutes per year of down time) the same
>as a Class 4/5 phone switch. This would require redundant
>design considerations in both hardware and software.

>>>
>>> To turn around, let's discuss what we need to focus on to get
>>> Asterisk there:
>>>
>>> Here's a few bullet points, there's certainly a lot more
>>> * Linux platform stability - how?
>>
>> Even more than Linux itself is the x86 platform... I've thought about
>> this a bit when considering * boxes for big customers.  When one
>> actually comes along, I'll have to actually make a decision :-).
>>>From where I stand, the best thing to do for smaller customers is give
>> them a box with RAID and redundant power supplies, if they can afford
>> it.
>
> You can overcome most of those problems by buying good quality
> hardware.  If you buy your * server from your local Taiwanese clone
> shop, you're asking for trouble.  A big, beefy machine from Dell would
> be better.

Yeah, but nothing like a nice, big Sun machine.  A cluster of Dell
machines is reliable, but a midrange Sun box puts them to shame.
>> But if I were to have a big customer with deep pockets, I'd really
>> like * on a big Sun beast with redundant-everything (i.e. you can hot
>> swap any component and there's usually n+1 of everything).  The
>> problem is that I don't think there's any Solaris support for Digium
>> cards, since it's kind of  a chicken-and-egg problem.
>
> Nope.  No Solaris support, but you might be able to get away with
> Linux/Solaris...but then you lose a lot of the hot-swapability.  In my
> experience, though, the only things I've ever been able to hotswap were
> power supplies and hard drives...and thats not software/os dependant.

With the big boxes like the 4800, you can hot swap CPUs and memory and
such as well.  You're right that all that stuff is pretty
Solaris-dependent, which is why I wanted to see if I couldn't get Asterisk
to run on a little Solaris machine (and then sell it to people who own the
big ones).
>> One of these days, I may convince myself to buy a modern Sun box
>> (maybe the ~$1000 Blade 100s) and see what can be done.  The only
>> problem I could conceive would be endian-ness, but I read about Digium
>> cards in a PowerPC box, so that won't be a problem, right?
>> Nick
>
> Endian-ness is really only a driver issue.  Its when programmers who
> believe that the world revolves around Linux/i386 that you have
> problems.

But it can also be a problem if you have on-card firmware, I've heard.

> Personally, I'd stick my Digium cards into an Alpha of some sort.  A
> DS-10L for 1U mounting with 1 card or a DS-20 for multiple cards where
> you need lots of processor zoobs.

I like the Alphas too, but they're being discontinued last I heard, and
being replaced with the Itanium.  Even VMS is being ported (now _there's_
an OS for * :-)
Nick


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Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)

[Nick Bachmann]

> Nick Bachmann wrote:
>
>>Yes, I've played with it a bit.  It's pretty simplistic... the
>>clustering just keeps several servers in sync with each other.  I
>>suppose that would be easy to do with Asterisk, especially if
>>configuration data was stored in a RDBMS that could do replication.
>>Even now, setting up a copy/reload routine isn't difficult.
>>It also seems that if you had a load balancer set up in front of your *
>>servers to balance the call requests, you'd have enough clustering to
>>keep one failure from taking down the whole system. Since the load
>>balancer keeps an affinity table (and monitors to make sure the servers
>>aren't going down) all VoIP connections could end up at the same * box
>>once they had been allocated, unless a server goes down, in which case
>>the call probably gets dropped. Any planned downtime could be made
>>without any disruptions, since you could stop the load balancer from
>>allocating any more connections to the * box and use 'stop when
>>convenient' to wait for all current calls to end.
>>Nick
>>
>>
>>
> As long as what ever system is used only presents a single IP address
> on  the network, the reason being that if a SIP UA is behind NAT the
> NAT  router will have opened a path for the response from the server it
>  contacted, if the request was offloaded to another IP address then the
>  response would not get through..

Yes, most load balancers have an "affinity table" (depending on your
vendor-speek) that pairs client and server IPs.
> Also the servers in the "cluster" would have to share SIP registration
> information so that all servers would know all availible UA's and all
> servers would have to communicate to that UA on the same IP address..

I forgot that... the VOCAL stuff also shares registration data.  It can't
be too hard for Asterisk to do this as well.
> These things could have major issues when it came to the RTP streams..

The affinity table makes the RTP stuff OK, but I agree that sharing SIP
registrations is a concern.
Nick


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Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)

[Nick Bachmann]

> On Sun, 2004-01-04 at 12:53, Nick Bachmann wrote:
>> Yes, I've played with it a bit.  It's pretty simplistic... the
>> clustering just keeps several servers in sync with each other.  I
>> suppose that would be easy to do with Asterisk, especially if
>> configuration data was stored in a RDBMS that could do replication.
>> Even now, setting up a copy/reload routine isn't difficult.
>
> A database doesn't make this easier. I would suggest you look into a
> revision control system and the ability to register
> applications/scripts to be run on check in of changes. Your benefit
> here is a quick roll out on hardware failure, plus roll back. You
> probably have seen people doing mailings based on CVS check ins, you
> could have those trigger a script on the clients that pulled fresh
> copies and did a reload. Fairly simple over all.

Yes, agree that CVS works well for this (that's how I manage my stuff).  I
like a RDBMSs for this kind of work, though, because the replication works
well and they are much faster than text files when you've got lots of
data.  Rolling back transactions is also pretty simple with most
databases, but I agree CVS is easier in this regard.
>> It also seems that if you had a load balancer set up in front of your
>> * servers to balance the call requests, you'd have enough clustering
>> to keep one failure from taking down the whole system. Since the load
>> balancer keeps an affinity table (and monitors to make sure the
>> servers aren't going down) all VoIP connections could end up at the
>> same * box once they had been allocated, unless a server goes down, in
>> which case the call probably gets dropped. Any planned downtime could
>> be made without any disruptions, since you could stop the load
>> balancer from allocating any more connections to the * box and use
>> 'stop when convenient' to wait for all current calls to end.
>
> The problem here is that you do have a single point of failure, the
> load balancer. It would be better to have multiple machines that you

That's why you buy a load balancer with its own redundancy :-).  The
Allied Telesyn SB series, for example, have two system controllers.  Cisco
stuff is probably similar.  The ATI stuff says about 30 seconds of
downtime when one SC fails, I would guess Cisco delivers less.
> selectively placed as primary and backup in your VoIP phones. It isn't
> true load balancing, but it does allow you only loose a specific amount
> of calls in progress at any time if a machine fails. Calls could then
> be picked up and restarted via the other machine. This would give fault
> tolerance, and would give the impression of having 5 9's as long as the
> failures are sufficiently spaced out.

I remind you that close only counts in horseshoes, hand grenades, and
nuclear weapons, and not with my users' uptime :-).
Nick


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Re: [Asterisk-Users] Hardware to build an Enterprise AsteriskUniversal Gateway

[Nick Bachmann]

> Andrew Kohlsmith wrote:
>>>I would set the "Enterprise Class" bar at five 9's reliability
>>>(about 5.25 minutes per year of down time) the same
>>>as a Class 4/5 phone switch. This would require redundant
>>>design considerations in both hardware and software.
>>
>
> To turn around, let's discuss what we need to focus on to get
> Asterisk there:
>
> Here's a few bullet points, there's certainly a lot more
> * Linux platform stability - how?

Even more than Linux itself is the x86 platform... I've thought about this
a bit when considering * boxes for big customers.  When one actually comes
along, I'll have to actually make a decision :-).
>From where I stand, the best thing to do for smaller customers is give
them a box with RAID and redundant power supplies, if they can afford it.
But if I were to have a big customer with deep pockets, I'd really like *
on a big Sun beast with redundant-everything (i.e. you can hot swap any
component and there's usually n+1 of everything).  The problem is that I
don't think there's any Solaris support for Digium cards, since it's kind
of  a chicken-and-egg problem.
One of these days, I may convince myself to buy a modern Sun box (maybe
the ~$1000 Blade 100s) and see what can be done.  The only problem I could
conceive would be endian-ness, but I read about Digium cards in a PowerPC
box, so that won't be a problem, right?
Nick



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Re: [Asterisk-Users] Asterisk, enterprise edition (New subject)

[Nick Bachmann]

> WipeOut wrote:
>
>>>
>> Asterisk would need some kind of clustering/load balancing ability
>> (Single IP system image for the IP phones across multiple servers) to
>> be  truely "Enterprise Class" in terms of both reliability and
>> scaleability..  Obviously that would not be as relevent for the analog
>>  hard wired phones unless the channelbanks and T1/E1 lines could be
>> automatically switched to another server..

Switching a T1 automagically seems like it would be an easy hack, but it
wouldn't be needed for customers who had more than one T1 (like, say, most
Enterprises :-).  The exception to this is people who are muxing their
internal phones, of course.
> Anyone that have peeked into Vovidas heartbeat/cluster architecture?

Yes, I've played with it a bit.  It's pretty simplistic... the clustering
just keeps several servers in sync with each other.  I suppose that would
be easy to do with Asterisk, especially if configuration data was stored
in a RDBMS that could do replication.  Even now, setting up a copy/reload
routine isn't difficult.
It also seems that if you had a load balancer set up in front of your *
servers to balance the call requests, you'd have enough clustering to keep
one failure from taking down the whole system. Since the load balancer
keeps an affinity table (and monitors to make sure the servers aren't
going down) all VoIP connections could end up at the same * box once they
had been allocated, unless a server goes down, in which case the call
probably gets dropped. Any planned downtime could be made without any
disruptions, since you could stop the load balancer from allocating any
more connections to the * box and use 'stop when convenient' to wait for
all current calls to end.
Nick


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Re: [Asterisk-Users] Grandstream Early Dial

[Nick Bachmann]

> On Thu, 18 Dec 2003, Aaron Martin wrote:
>
>> I have upgraded my grandstream phone from firmware 1.0.3.78 to
>> 10.0.4.30 and now I am having problems with early dial.  On the older
>> firmware earlydial worked fine with my asterisk server, but now as
>> soon  as I have dialed the number I get a congested tone, and the
>> number 4  flashes up on the LCD screen.
>>
>> Has anyone had this problem, and if so, how do I fix it?
>
> Early dial has never worked for me, and I just upgraded to the 1.0.4.30
>  load yesterday. Now, I am having DTMF recognition issues, making it
> impossible to check my voice mail.

Are you using SIP Info for DTMF?  It's the only thing that reliably works
with GS phones.
> Not sure what to do, but I'm hoping that my SNOM 200 IP Phone will
> yield  better results.

It almost certainly will.  (Some would say that two tin cans and a string
would work better than Grandstream phones, but I digress...)
Nick


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Re: [Asterisk-Users] CVS Closed?

[Nick Bachmann]

Carl A. Cook wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Thanks, to the two.

But I can't tell you how weary and sick I am of this kind of reflexive
criticism.  It is -always- counter to true advancement, and burns those
delicate few who actually cause progress, in addition to (pitiable) n00bs.
I run a small new architecture company, and am a vocal open standards
advocate.  I'm an academic, a free-thinker, and researcher, and the thought
of investing the large amount of effort and time required to learn a new
thing in detail isn't a problem;  but if it's necessary to join another
bummed-out Thunderdome, where countless bloody-minded fukkers are sitting in
the bushes waiting to mean-spiritedly bite my ass off whenever I ask a stupid
One thing that is very hard for people new to the Open Source 
development world is the idea that the people on the mailing list (or 
IRC channel) don't have to be nice to you.  This is not Dell; we're not 
your wet blanket of support -- we don't get paid to be. (If you want 
that kind of thing, email me off list, I run a consulting company where 
we'll do that for you if you give us money :-) ) You're just not used to 
getting reproached by tech support.  Accept that you asked a stupid 
question and got called on it.  It's not like we're going to hold it 
against you (unless, of course, you become a compulsive 
dumb-question-asker).

That said, I've noticed that most of the people on this list are usually 
very helpful if you do your part in the whole ask-questions-get-answers 
scheme.  Go read http://www.catb.org/~esr/faqs/smart-questions.html and 
then, hopefully, you'll feel better about what happened now.

question or have a 'different' idea, I can't spare the resources.

I don't think you'll have trouble with different ideas: people here are 
open to change.  If you lurk around a bit, you'll see that this is a 
very open community.

[Deeper meaning of existance snipped]

That said, I'm trying very hard to swallow this hairball, and investigate
asterisk.
 

So basically, you did something dumb: build a bridge and get over it. 
Go read the documentation: there's a book at 
http://www.asteriskdocs.org and a handbook, along with a wiki at 
voip-info.org and lots of good web pages.  When you coalesce the 
information they contain, you'll find that they have the answers to most 
of your questions.

Nick

P.S. Don't forget to read 
http://www.catb.org/~esr/faqs/smart-questions.html.

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Re: [Asterisk-Users] Help with x101P

[Nick Bachmann]

Andrew Thompson wrote:

- Original Message -
From: "Burak Balasaygun" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, December 27, 2003 10:46 PM
Subject: RE: [Asterisk-Users] Help with x101P

 

I'm not sure what you mean by what type of switch you are connected to?
   

The
 

x101p is connected to the CO switch for my LEC.

   

Occasionally I do NPA-NXX lookups for my local exchanges to see what other
carriers have prefixes in my area. I used to use telcodata.us, but they seem
to have gone offline. Usually, after you find the carrier's name, you can
see info on the location and type of switch being used. I can't say with any
assurity that the info is accurate, but it is there if you dig.
 

http://www.telcoexchange.com/resources/carriers/index.shtml will tell 
you the switch and other stuff when you put in a phone number.

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Re: [Asterisk-Users] Grandstream Quality Survey.... :P

[Nick Bachmann]

Brian West wrote:

Today class we are going to be talking about the wonderful line of
GrandStream products.  Or should I say BarbieTone phones?
 

OK, so GrandStream phones are crap.  What other phone products are there 
on the market that are cheap (and I DO NOT want to buy phones off eBay 
for a business), but work well.  ATAs are out of the question because 
they aren't phones... and don't support all possible VoIP features.

Cisco phones are all at least $200+, with maintenance, right?

So what other options are there for ~$100 SIP/IAX hardphones?

Nick

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Re: [Asterisk-Users] 911 settings.

[Nick Bachmann]

Andrew Thompson wrote:

- Original Message -
From: "Ariel Batista" <[EMAIL PROTECTED]>
To: "Asterisk User List" <[EMAIL PROTECTED]>
Sent: Friday, December 19, 2003 4:06 PM
Subject: [Asterisk-Users] 911 settings.
 

I would like to know if anyone has come up with a script for 911 dialing
rules that put correct information on our locations.  We have our office
in 3 different building one being our production & shipping dock.  It is
almost 2 blocks away.  We are connected with Ethernet Wireless between
the buildings and have Sip phones setup in the other 2 locations. All
the phones are working just fine.  But when they call 911 they get our
main address and not the other address's.  So we need to be able to give
the correct address to the 911 call!  This is just for our locations and
not for reselling our Asterisk server!
   



The 911 office is most likely retreiving the address off of the line that is
placing the call. Do you have any voice lines in the other buildings?
I would consider a line siege device and FXO attached to a fax or security
system line in the other buildings. Route the dialed 911's out over the
local pots line and they will get the correct address. I don't know if you
can attach an address any other way.
You could try sending a different callerid, but if they are all billed as
being in the main building, that's probably the address they'll get.
 

Another solution that might work is to ask the phone company to change 
the address that they give the PSAP on one of your phone numbers to the 
other building, and then use that for 911.  

I don't know how big of a customer you are for your phone company, but 
if you have more than a token number of lines they'll hopefully go for it.



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Re: [Asterisk-Users] Manager API Problem

[Nick Bachmann]

Michael Devenijn wrote:

Everythings works great with asterisk exept one feature with redirect 
: it doesn't redirect when ringing ...
Have you used astman with a new CVS?  It works for me...

If not, you'll need to post more information for the list to help you.

Nick

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Re: [Asterisk-Users] Asterisk in a Centrex environment?

[Nick Bachmann]

Peter Pauly wrote:

Does anyone know what would be involved in making
Asterisk work as a voicemail system in a Centrex 
environment?  We have a Centrigram voicemail system
that belongs in the Smithsonian. There are analog
lines coming into the box and a 56KB data feed from
the phone company's switch. 
 

Note that I'm no expert on this kind of configuration, but this sounds 
pretty easy.  Get a channel bank (or better yet, just get a T1 from the 
ILEC) and a Digium card. Configuring voicemail and extensions is pretty 
easy, just look at voicemail.conf.  The only puzzling part is the data 
feed... what's it for, determining which line is ringing which voicemail 
box?  If so, perhaps it can be converted to DNIS on the T1?

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Re: [Asterisk-Users] (no subject)

[Nick Bachmann]

Adam Hart wrote:

Hence why I ask for a company name. Small correction to your post, if it's
distributed to anyone, the source must be available to EVERYONE.
IANAL, but I don't think that's quite accurate.  If this person wanted 
to, they could only ofter an offer for the source to people who bought 
the software from them.  The receiver of the software can then, if he or 
she chooses to, give that offer to anyone they want, and it must be 
offered.  But without an ofter, the seller is not obligated to give you 
the source.  From the GPL FAQ:

What does this "written offer valid for any third party" mean? Does that 
mean everyone in the world can get the source to any GPL'ed program no 
matter what? 

"Valid for any third party" means that anyone who has the offer is 
entitled to take you up on it.

If you commercially distribute binaries not accompanied with source 
code, the GPL says you must provide a written offer to distribute the 
source code later. When users non-commercially redistribute the binaries 
they received from you, they must pass along a copy of this written 
offer. This means that people who did not get the binaries directly from 
you can still receive copies of the source code, along with the written 
offer.

The reason we require the offer to be valid for any third party is so 
that people who receive the binaries indirectly in that way can order 
the source code from you.


- Original Message - 
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 5:48 PM
Subject: Re: [Asterisk-Users] (no subject)

Well if it links to asterisk and or used any of its code as a base it
can't be sold without a comercial lic. for asterisk.  Thats my
understanding of the GPL.  If its sold then all the source has to go along
with it right?
bkw

On Tue, 9 Dec 2003, Adam Hart wrote:

 

Is there a company website? or just a free yahoo email address?

- Original Message -
From: "Kita B. Ndara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, December 09, 2003 4:01 PM
Subject: [Asterisk-Users] (no subject)
   

Hi,

Our firm has developed two applications that I
thought might be of interest to members of this list
as both run over Asterisk:
The first is a calling card application that covers
needs in that area: scratch number generation, call
termination via least-cost route (i.e. multiple
termination providers), etc.  We have tested this with
voicepulse as our termination provider and it works
great.
The second is a call centre system: Call queueing,
distribution, real-time reporting, statistics.
Backend database is PostgreSQL (with pgcrypto module)
for both applications, and in keeping with the
Asterisk spirit, call origin/destination is h/w and
software independent.
If anybody is interested in these, please contact me
off-list and I'll be happy to discuss these with you.
Thanks

B.


BT Yahoo! Broadband - Save £80 when you order online today. Hurry! Offer
 

ends 21st December 2003. The way the internet was meant to be.
http://uk.rd.yahoo.com/evt=21064/*http://btyahoo.yahoo.co.uk
   

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Re: [Asterisk-Users] Asterisk Maint.

[Nick Bachmann]

> What kind of stability / reliability are people currently experiencing
> with the Linux / Asterisk combination?  We will be running 3-10 SIP
> phones from India to US using nothing more than regular cable / dsl
> connections from both locations.

People have had months of uptime.  I would be more concerned with the
reliability of your DSL/Cable reliability.  You should also use a
Bandwidth calculator (like http://www.packetizer.com/iptel/bandcalc.html)
to figure out how much bandwidth you're going to need and compare that to
how much will be available.
> Also, what make / model SIP phone do you recommended that would allow
> us to configure the phones to work on alternate ports (or is this a
> standard configuration option on most SIP phones) ?

Most do.  I would reccomend Grandstream phones for your application,
because they're cheap, easy, and tested with Asterisk.
Nick


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Re: [Asterisk-Users] grandstream budgeTone phones or Asterisk ??

[Nick Bachmann]

Senad Jordanovic wrote:

Nick Bachmann wrote:
 

Hello

I have couple of Grandstream phone and some of them after a day or
two just stops receiving calls, you can still make a call from that
phone but you cannot receive calls until you restart the phone. Is
it a wrong configuration of phone or Asterisk ? Thanks for any
advices. 
 

Try lowering their reregistering time.  I had a similar problem and I
discovered (with sip show peers) that Asterisk no longer had their IP
stored.  Lowering the reregister interval seemed to fix the problem.
Nick   

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Hi,
To what figure did you lower it?
 

Two minutes, just for testing.  That seemed to take care of it, but 
perhaps it prevented a true cause from being found.

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Re: [Asterisk-Users] Replicating Legacy Phone Behavior

[Nick Bachmann]

Jonathan Moore wrote:
>
> Quoting Nick Bachmann <[EMAIL PROTECTED]>:

>> Having a DSS (the blinking lights for each extension, short for
>> "Digital Station Selector") is a feature that I wish Asterisk had.  A
>> week or so ago there was discussion about a new Windows-based Asterisk
>> application (Asterisk Call Manager for Windows?) and it was said that
>> in a later version there was a plan to add a "Console mode" (the name
>> for the Uberphone that the DSS attaches to).  If I weren't so swamped,
>> I'd ofter to help out :-).
>>
>> Figuring out which extensions were busy would be easy with the Manager
>> API, but I'm not sure how you could forward incoming calls bound for
>> another extension.  I guess if it were easy enough, I guess I could
>> mke  a Javax/Swing app to do it.
>>
>> If there's already an app that does this, I haven't see it, but I'd
>> love to!
>>
>
> I think this is a hard one to deal with because you are going to need
> support  for this on both the phone and in *. I am really know expert

My thought was a Java app that just transfered to your phone.  So if a
line rings and someone is DND/unavailable anther person can just forward
it to their cheapo phone.  I think "redirect" might do that, but I've yet
to expirament.
> but I think the  only phones on the market that might be turned into
> this are some of the  programmable ones, maybe the 7690 or the Pingtel
> by programming the "lights" on  the lcd. Pingtels look to have a nice
> display on them and are supposed to be  programmable with Java. The
> problem I see with this is that these are all high  end fairly
> expensive phones. The snom 200 may have some hooks for this, but are
> limited to the 5 programmable buttons.
>
> The more I think about it the more I think the 7960 might be doable,
> since I  know my Cisco sales rep was trying to sell me on DSS.

I've seen those but I'm pretty sure you have to use CCM to use them.

>
> I also remember reading some references to SIP protocol updates that
> might  include some of these types of features (pageing also)
>
>> >2. How does one go about creating call queues and advanced features
>> >such  as UCD and ACD using Asterisk?
>> >
>> Take a look at
>> http://www.voip-info.org/wiki-Asterisk+config+queues.conf and the
>> pages it references.
>>
>> >3. Is it possible to do Phone to Phone paging with SIP phones? This
>> >is a  feature that I personally use a lot on my Legacy Phone System.
>> >I simply  hit the extension of the persion I want to chat, and it
>> >beeps their phone  and we can talk. Sort of like an Intercom system.
>> >
>>
>> That would be a phone feature... I think some of the Cisco phones do
>> it... it's billed as "AutoAnswer" I think.
>>
>
> I have been looking at this angle too. I think the trick is to find a
> phone  that is a "multi line" VoiP model and allows per line
> configuration of auto- answer. I was thinking of using a pattern where
> even extensions are for ringing  the phone and odd numbers are for the
> intercom/pageing. Candidate phones that  may be able to do this that I
> am researching include snoms, cisco 7960, and  swiss something (mcgp
> based phone). I have mostly been looking at the low end  of the phone
> market, so there may be many others at the high end I am not aware  of.

I saw on the list a while back somebody figured out how to do this with a
Cisco instrument. It's not a big deal do me, as none of the PBXs in our
buildings do this, thus it's not a feature our staff has decided they
can't live without.

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Re: [Asterisk-Users] grandstream budgeTone phones or Asterisk ??

[Nick Bachmann]

> Hello
>
> I have couple of Grandstream phone and some of them after a day or two
> just stops receiving calls, you can still make a call from that phone
> but you cannot receive calls until you restart the phone.
> Is it a wrong configuration of phone or Asterisk ?
> Thanks for any advices.

Try lowering their reregistering time.  I had a similar problem and I
discovered (with sip show peers) that Asterisk no longer had their IP
stored.  Lowering the reregister interval seemed to fix the problem.
Nick


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Re: [Asterisk-Users] Replicating Legacy Phone Behavior

[Nick Bachmann]

Greg Boehnlein wrote:

	First and foremost, these Key System installers are big believers 
in VoIP and convergence technologies. While the KSU vendors may see 
This has been my experiance as well.  Everybody but PBX vendors like 
VoIP.  The KSU people like it because it gives them more work and job 
security, but the vendors don't like protocols they don't allow them to 
have lock-in.  After all, I can use a mix of phones on my * PBX, but if 
I want a digital phone on my Fujitsu system, I've got to get a Fuji 
phone.  Phones can be like razor blades for Norelco... it's the product 
that keeps on selling itself!

Asterisk as competition, the installers on the ground see it as an 
excellent addition to help connect remote offices and workers together, 
but they are driven by the needs of their customers, most of whom want to 
KISS (Keep It Simple, Stupid). I.E. they want an Asterisk based VoIP 
solution to work in a similar manner to their existing PBX or Phone 
System
>
	As a result, these are some of the questions that they threw at me 
that I am trying to figure out:

1. Legacy KSU and PBX users are used to seeing blinking lights on their 
phone that indicate outside lines in use, call on hold, voice mail 
waiting, do not disturb etc.. Is it possible to have these features using 
SIP phones on the dekstop? I.E. if a user puts a caller on hold at one 
extension, can it blink a light on all extensions so that user can be 
picked up at another extension? This gets into issues regarding 
re-training people with new phones etc.. Kind of like the issue of "I 
don't want to press enter to make a call.. Why can't this phone just work 
like my old analog phone?"
 

Having a DSS (the blinking lights for each extension, short for "Digital
Station Selector") is a feature that I wish Asterisk had.  A week or so
ago there was discussion about a new Windows-based Asterisk application
(Asterisk Call Manager for Windows?) and it was said that in a later
version there was a plan to add a "Console mode" (the name for the
Uberphone that the DSS attaches to).  If I weren't so swamped, I'd ofter
to help out :-).
Figuring out which extensions were busy would be easy with the Manager
API, but I'm not sure how you could forward incoming calls bound for
another extension.  I guess if it were easy enough, I guess I could mke 
a Javax/Swing app to do it.

If there's already an app that does this, I haven't see it, but I'd love to!

2. How does one go about creating call queues and advanced features such 
as UCD and ACD using Asterisk?

Take a look at http://www.voip-info.org/wiki-Asterisk+config+queues.conf
and the pages it references.
3. Is it possible to do Phone to Phone paging with SIP phones? This is a 
feature that I personally use a lot on my Legacy Phone System. I simply 
hit the extension of the persion I want to chat, and it beeps their phone 
and we can talk. Sort of like an Intercom system.
 
That would be a phone feature... I think some of the Cisco phones do 
it... it's billed as "AutoAnswer" I think.

Nick

P.S. The Asterisk-users lists are searchable if you want to check if 
your question has been answered before you post.  Also, the 
voip-info.org Wiki is very informative.

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Re: [Asterisk-Users] Sip phones!

[Nick Bachmann]

Ariel Batista wrote:

I am trying to get the following phones for testing.  Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters?  I can not afford the Cisco phones there too hard to configure and too expensive!

1 - Sipura SPA-2000
2 - Grandstream Sip phone BT-102
Grandstream told me to go to Ovislink 
(http://www.ovislink.com/newovislink/IPPhones.asp) to get phones. 
They're one of the only US places I've been able to find BT-102s.

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Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)

[Nick Bachmann]

Steven Sokol wrote:

I have looked at creating a "Console" version of the application.  It
would be very much like a DSS ("Direct Station Selector" for the
non-AT&T/Avaya initiated).  It would support either click-to-transfer or
drag-and-drop transfer of incoming calls. 

Excellent!  This is one feature we really need!  The way I thought about 
doing this would be to stick the console as an icon in the System Tray 
(where the volume control, et. al. is) and have it pop up a window 
(similar to Gnome's calender/date applet) whenever a call comes in that 
must be brought to the person's attention.  For example, if Bob's 
secretary wasn't answering her phone (or had marked herself as away), 
the console would pop up for all the other secretaries with the incoming 
call.  Since they'd know the extension, it would be transparent to the 
caller that it was really Jane's secretary answering.

May I suggest you use something like HBasic -- it's like Visual Basic,
but can be used on Windows and Linux.
   

HBasic?  Cool.  I have had somebody else suggest wxWindows as a method
of building the GUI.  Do the two play nicely together?
 

Well, HBasic uses Qt for its GUI and wxWindows is a GUI library for 
C/C++ (and Perl, and...) similar to Qt.  So basically, they're apples 
and oranges.

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Re: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha)

[Nick Bachmann]

Steven Sokol wrote:

1. Redial
2. Voicemail Box Monitoring
3. Enhanced Conferencing
4. Outlook/Act/Goldmine Integration (PIM stuff)
5. Call History (both inbound and outbound)
6. Redirect Option on Ring (VM, Application, Transfer, etc.)
7. Automatic mixing and delivery of monitored (recorded) files.
 

What would be neat would be a limited setup for secretaries or 
receptionists to be able to see incoming calls on all extensions and be 
able to forward them to thier phones... like a DSS.  Maybe there's a way 
to do it now, but not that I know of.

A copy of the source code (let's call this LGPL for now) is available
here:
http://www.sokol-associates.com/Downloads/AstMgr.zip
It's written in VB6 (yes - barf, gag, whatever).  The only thing
required beyond the integral VB6 controls is the Windows Scripting
Runtime which most PCs should have.  I will work on an installable
version soon.  I may also port it to something more cross-platform.
Please bear with me as I am just learning Gnome/GTK/X-windows.
 

May I suggest you use something like HBasic -- it's like Visual Basic, 
but can be used on Windows and Linux.

Nick

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