Re: [asterisk-users] Problems Solved, two left

[Daryl Richards]

On 2023-05-23 7:22 p.m., Steve Matzura wrote:

And I think they're both small.


[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: 
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because 
extension not found in context 'voipms-inbound'.


Steve,

In your voip.ms console, go to Account Settings -> Inbound Settings, and 
set Device Type to "IP PBX Server..." instead of "ATA device..."


This will fix the 's' instead of the number.



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Re: [asterisk-users] Metasphere?

[Daryl Jones]

On 3/25/2010 8:13 AM, David Gibbons wrote:
 Hi All

 I'm involved in discussions with my carrier right now and am wondering if 
 anyone has interconnected Asterisk to Metasphere via SIP?
   


Yes, we're served by a Metaswitch usng SIP.  Works fine.

-Daryl



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Re: [asterisk-users] IAX complaints? What are they?

[Daryl G. Jurbala]

How recent?  I tried switching from 1.2 to 1.4 about 4 months ago, and  
asterisk would stop accepting IAX connections in less than a day and  
would need to be restarted.

This is with about 50 to 100 calls at a time on each box for about 10  
or 12 hours a day.  Less for the other half.  And all IAX calls are  
being passed on to a far end terminator via SIP.

I was going to scrap IAX entirely because it didn't seem to scale well  
(for non-trunking apps, at least), but many customers need it for  
various reasons.
Daryl

On Nov 30, 2007, at 8:52 AM, zoa wrote:

 IAX had some stability issues in the past, the recent releases have a
 lot of iax2 fixes and should no longer have those issues.

 Zoa


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[asterisk-users] Set CALLERID(num) to a specific number only if ${CALLERID(num)} is not an NANP number

[Daryl G. Jurbala]

Im trying to figure out the base way to check the callerID being sent  
to my Asterisk box and use it if it is a valid NANP number, but  
replace it with a static NANP number if it is not.  (Why?  I have a  
few carriers that require this, and a few international users - if it  
happens to take one of the carriers that require it, I want it to set  
a static number that is valid).

I'm playing with IF and REGEX in extensions.conf, but not getting  
very far.  Has anyone done this and/or know of a doc?  I haven't had  
any success searching.

At this point, I have a very broken setup of:

Set(CALLERID(num)=${IF(${REGEX(^(?:\([2-9]\d{2}\)\ ?|[2-9]\d{2}(?: 
\-?|\ ?))[2-9]\d{2}[- ]?\d{4}$ ${CALLERID(num)})}?${CALLERID 
(num)}:staticNumber)

I'm sure I'm pretty far off - and I've been through many permutations  
of this so far.  Any ideas?

Thanks,
Daryl

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Re: [asterisk-users] AGI answering the channel even though I neverasked it to

[Daryl G. Jurbala]

On Aug 13, 2007, at 4:37 PM, Martin Smith wrote:

 See
 http://www.asterisk.org/doxygen/1.4/ 
 res__agi_8c.html#c631d48f46d51d4b057
 b31807baa1f10

 The AGI application will answer the channel if it isn't already
 answered.

 You probably need to do whatever you want to do in the dialplan, and
 keep using DeadAGI.


Excellent information.  That's what I spent an hour or so  
unsuccessfully looking for ;)

Thank you very much.

Now I just have to figure out how to do a database lookup without  
answering the channel, as that seems to indicate that the AGI is  
going to answer regardless of whether a play progress tones or not  
from the AGI.
Daryl

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[asterisk-users] AGI answering the channel even though I never asked it to

[Daryl G. Jurbala]

I am working on a call-back solution where the initiating call should  
never be answered.

I was doing this simply through the dial plan, sending a progress  
tone, and then dumping the channel, and firing off a DeadAGI which  
created a call file to make the callback.

Now I've tried extending this so that an AGI is fired first to check  
for things - like no inbound ANI - and play a DIFFERENT progress tone  
for that situation.  It appears that every since I've done that,  
Asterisk is answering the channel.  I don't have an Answer command in  
my dialplan or AGI.  Is this something that will automagically happen  
whether I want it to or not?  If so, I'm going to have to do some  
ugly dial plan scripting to make this work.

In case it matters, this is a PHP AGI.

Thanks,
Daryl

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[asterisk-users] Problem building Asterisk 1.2.22

[Daryl Jones]

I'm having a problem building Asterisk 1.2.22. It fails in 
codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.

Here's the error. Can anyone help me with this?

gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
-Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT 
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS 
-fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c
codec_zap.c: In function ‘zap_framein’:
codec_zap.c:143: error: dereferencing pointer to incomplete type
codec_zap.c:145: error: dereferencing pointer to incomplete type
codec_zap.c:147: error: dereferencing pointer to incomplete type
codec_zap.c:147: error: dereferencing pointer to incomplete type
codec_zap.c:152: error: dereferencing pointer to incomplete type
codec_zap.c:152: error: dereferencing pointer to incomplete type
codec_zap.c:152: error: dereferencing pointer to incomplete type
codec_zap.c:154: error: dereferencing pointer to incomplete type
codec_zap.c:154: error: dereferencing pointer to incomplete type
codec_zap.c:154: error: dereferencing pointer to incomplete type
codec_zap.c:154: error: dereferencing pointer to incomplete type
codec_zap.c:155: error: dereferencing pointer to incomplete type
codec_zap.c:158: error: dereferencing pointer to incomplete type
codec_zap.c:158: error: dereferencing pointer to incomplete type
codec_zap.c:158: error: dereferencing pointer to incomplete type
codec_zap.c:159: error: dereferencing pointer to incomplete type

codec_zap.c: In function ‘zap_frameout’:
codec_zap.c:183: error: dereferencing pointer to incomplete type
codec_zap.c:192: error: dereferencing pointer to incomplete type
codec_zap.c:193: error: dereferencing pointer to incomplete type
codec_zap.c:194: error: dereferencing pointer to incomplete type
codec_zap.c:194: error: dereferencing pointer to incomplete type
codec_zap.c:195: error: dereferencing pointer to incomplete type
codec_zap.c:196: error: dereferencing pointer to incomplete type
codec_zap.c:199: error: dereferencing pointer to incomplete type
codec_zap.c:202: error: dereferencing pointer to incomplete type
codec_zap.c:203: error: dereferencing pointer to incomplete type
codec_zap.c:204: error: ‘ZT_TCOP_TRANSCODE’ undeclared (first use in 
this function)
codec_zap.c:204: error: (Each undeclared identifier is reported only once
codec_zap.c:204: error: for each function it appears in.)
codec_zap.c:205: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this 
function)
codec_zap.c: In function ‘zap_destroy’:
codec_zap.c:219: error: ‘ZT_TCOP_RELEASE’ undeclared (first use in this 
function)
codec_zap.c:220: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this 
function)
codec_zap.c:223: error: dereferencing pointer to incomplete type
codec_zap.c: In function ‘zap_new_alawtog723’:
codec_zap.c:240: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this 
function)
codec_zap.c:262: error: dereferencing pointer to incomplete type
codec_zap.c:269: error: dereferencing pointer to incomplete type
codec_zap.c:269: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in 
this function)
codec_zap.c:270: error: dereferencing pointer to incomplete type
codec_zap.c:271: error: dereferencing pointer to incomplete type
codec_zap.c:277: error: dereferencing pointer to incomplete type
codec_zap.c:278: error: dereferencing pointer to incomplete type
codec_zap.c:279: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this 
function)
codec_zap.c:281: error: dereferencing pointer to incomplete type
codec_zap.c: In function ‘zap_new_ulawtog723’:
codec_zap.c:297: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this 
function)
codec_zap.c:319: error: dereferencing pointer to incomplete type
codec_zap.c:326: error: dereferencing pointer to incomplete type
codec_zap.c:326: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in 
this function)
codec_zap.c:327: error: dereferencing pointer to incomplete type
codec_zap.c:328: error: dereferencing pointer to incomplete type
codec_zap.c:334: error: dereferencing pointer to incomplete type
codec_zap.c:335: error: dereferencing pointer to incomplete type
codec_zap.c:336: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this 
function)
codec_zap.c:338: error: dereferencing pointer to incomplete type
codec_zap.c: In function ‘zap_new_g723toalaw’:
codec_zap.c:354: error: ‘ZT_TCOP_ALLOCATE’ undeclared (first use in this 
function)
codec_zap.c:376: error: dereferencing pointer to incomplete type
codec_zap.c:383: error: dereferencing pointer to incomplete type
codec_zap.c:383: error: ‘ZT_TRANSCODE_MAGIC’ undeclared (first use in 
this function)
codec_zap.c:384: error: dereferencing pointer to incomplete type
codec_zap.c:385: error: dereferencing pointer to incomplete type
codec_zap.c:391: error: dereferencing pointer to incomplete type
codec_zap.c:392: error: dereferencing pointer to incomplete type
codec_zap.c:393: error: ‘ZT_TRANSCODE_OP’ undeclared (first use in this 
function)
codec_zap.c:395: error: dereferencing 

Re: [asterisk-users] Problem building Asterisk 1.2.22

[Daryl Jones]

Correct.  zaptel-1.2.12 is currently installed.  I plan to install 
zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but 
has not been installed yet.

John covici wrote:
 I wonder what version of Zaptel you are using -- sounds like you have
 not installed a new version or you are using an older one.

 on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
   I'm having a problem building Asterisk 1.2.22. It fails in 
   codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
   
   Here's the error. Can anyone help me with this?
   
   gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
   -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT 
   -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS 
   -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c
   codec_zap.c: In function ?zap_framein?:
   codec_zap.c:143: error: dereferencing pointer to incomplete type
   codec_zap.c:145: error: dereferencing pointer to incomplete type
   codec_zap.c:147: error: dereferencing pointer to incomplete type
   codec_zap.c:147: error: dereferencing pointer to incomplete type
   codec_zap.c:152: error: dereferencing pointer to incomplete type
   codec_zap.c:152: error: dereferencing pointer to incomplete type
   codec_zap.c:152: error: dereferencing pointer to incomplete type
   codec_zap.c:154: error: dereferencing pointer to incomplete type
   codec_zap.c:154: error: dereferencing pointer to incomplete type
   codec_zap.c:154: error: dereferencing pointer to incomplete type
   codec_zap.c:154: error: dereferencing pointer to incomplete type
   codec_zap.c:155: error: dereferencing pointer to incomplete type
   codec_zap.c:158: error: dereferencing pointer to incomplete type
   codec_zap.c:158: error: dereferencing pointer to incomplete type
   codec_zap.c:158: error: dereferencing pointer to incomplete type
   codec_zap.c:159: error: dereferencing pointer to incomplete type
   
   codec_zap.c: In function ?zap_frameout?:
   codec_zap.c:183: error: dereferencing pointer to incomplete type
   codec_zap.c:192: error: dereferencing pointer to incomplete type
   codec_zap.c:193: error: dereferencing pointer to incomplete type
   codec_zap.c:194: error: dereferencing pointer to incomplete type
   codec_zap.c:194: error: dereferencing pointer to incomplete type
   codec_zap.c:195: error: dereferencing pointer to incomplete type
   codec_zap.c:196: error: dereferencing pointer to incomplete type
   codec_zap.c:199: error: dereferencing pointer to incomplete type
   codec_zap.c:202: error: dereferencing pointer to incomplete type
   codec_zap.c:203: error: dereferencing pointer to incomplete type
   codec_zap.c:204: error: ?ZT_TCOP_TRANSCODE? undeclared (first use in 
   this function)
   codec_zap.c:204: error: (Each undeclared identifier is reported only once
   codec_zap.c:204: error: for each function it appears in.)
   codec_zap.c:205: error: ?ZT_TRANSCODE_OP? undeclared (first use in this 
   function)
   codec_zap.c: In function ?zap_destroy?:
   codec_zap.c:219: error: ?ZT_TCOP_RELEASE? undeclared (first use in this 
   function)
   codec_zap.c:220: error: ?ZT_TRANSCODE_OP? undeclared (first use in this 
   function)
   codec_zap.c:223: error: dereferencing pointer to incomplete type
   codec_zap.c: In function ?zap_new_alawtog723?:
   codec_zap.c:240: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this 
   function)
   codec_zap.c:262: error: dereferencing pointer to incomplete type
   codec_zap.c:269: error: dereferencing pointer to incomplete type
   codec_zap.c:269: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in 
   this function)
   codec_zap.c:270: error: dereferencing pointer to incomplete type
   codec_zap.c:271: error: dereferencing pointer to incomplete type
   codec_zap.c:277: error: dereferencing pointer to incomplete type
   codec_zap.c:278: error: dereferencing pointer to incomplete type
   codec_zap.c:279: error: ?ZT_TRANSCODE_OP? undeclared (first use in this 
   function)
   codec_zap.c:281: error: dereferencing pointer to incomplete type
   codec_zap.c: In function ?zap_new_ulawtog723?:
   codec_zap.c:297: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in this 
   function)
   codec_zap.c:319: error: dereferencing pointer to incomplete type
   codec_zap.c:326: error: dereferencing pointer to incomplete type
   codec_zap.c:326: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in 
   this function)
   codec_zap.c:327: error: dereferencing pointer to incomplete type
   codec_zap.c:328: error: dereferencing pointer to incomplete type
   codec_zap.c:334: error: dereferencing pointer to incomplete type
   codec_zap.c:335: error: dereferencing pointer to incomplete type
   codec_zap.c:336: error: ?ZT_TRANSCODE_OP? undeclared (first use in this 
   function)
   codec_zap.c:338: error: dereferencing pointer to incomplete type
   codec_zap.c: In function ?zap_new_g723toalaw?:
   codec_zap.c:354: error: ?ZT_TCOP_ALLOCATE? undeclared (first

Re: [asterisk-users] Problem building Asterisk 1.2.22

[Daryl Jones]

That's what I needed to know. Thanks!


John covici wrote:
 But asterisk will not compile till you install the correct version of
 zaptel.

 on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
   Correct.  zaptel-1.2.12 is currently installed.  I plan to install 
   zaptel-1.2.19 as part of this upgrade. zaptel-1.2.19 compiled clean, but 
   has not been installed yet.
   
   John covici wrote:
I wonder what version of Zaptel you are using -- sounds like you have
not installed a new version or you are using an older one.
   
on Wednesday 07/18/2007 Daryl Jones([EMAIL PROTECTED]) wrote
  I'm having a problem building Asterisk 1.2.22. It fails in 
  codecs/codec_zap.c on codec_zap.c is revision 62173. The OS is FC4.
  
  Here's the error. Can anyone help me with this?
  
  gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes 
  -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT 
  -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS 
  -fomit-frame-pointer -fPIC -c -o codec_zap.o codec_zap.c
  codec_zap.c: In function ?zap_framein?:
  codec_zap.c:143: error: dereferencing pointer to incomplete type
  codec_zap.c:145: error: dereferencing pointer to incomplete type
  codec_zap.c:147: error: dereferencing pointer to incomplete type
  codec_zap.c:147: error: dereferencing pointer to incomplete type
  codec_zap.c:152: error: dereferencing pointer to incomplete type
  codec_zap.c:152: error: dereferencing pointer to incomplete type
  codec_zap.c:152: error: dereferencing pointer to incomplete type
  codec_zap.c:154: error: dereferencing pointer to incomplete type
  codec_zap.c:154: error: dereferencing pointer to incomplete type
  codec_zap.c:154: error: dereferencing pointer to incomplete type
  codec_zap.c:154: error: dereferencing pointer to incomplete type
  codec_zap.c:155: error: dereferencing pointer to incomplete type
  codec_zap.c:158: error: dereferencing pointer to incomplete type
  codec_zap.c:158: error: dereferencing pointer to incomplete type
  codec_zap.c:158: error: dereferencing pointer to incomplete type
  codec_zap.c:159: error: dereferencing pointer to incomplete type
  
  codec_zap.c: In function ?zap_frameout?:
  codec_zap.c:183: error: dereferencing pointer to incomplete type
  codec_zap.c:192: error: dereferencing pointer to incomplete type
  codec_zap.c:193: error: dereferencing pointer to incomplete type
  codec_zap.c:194: error: dereferencing pointer to incomplete type
  codec_zap.c:194: error: dereferencing pointer to incomplete type
  codec_zap.c:195: error: dereferencing pointer to incomplete type
  codec_zap.c:196: error: dereferencing pointer to incomplete type
  codec_zap.c:199: error: dereferencing pointer to incomplete type
  codec_zap.c:202: error: dereferencing pointer to incomplete type
  codec_zap.c:203: error: dereferencing pointer to incomplete type
  codec_zap.c:204: error: ?ZT_TCOP_TRANSCODE? undeclared (first use in 
  this function)
  codec_zap.c:204: error: (Each undeclared identifier is reported only 
 once
  codec_zap.c:204: error: for each function it appears in.)
  codec_zap.c:205: error: ?ZT_TRANSCODE_OP? undeclared (first use in 
 this 
  function)
  codec_zap.c: In function ?zap_destroy?:
  codec_zap.c:219: error: ?ZT_TCOP_RELEASE? undeclared (first use in 
 this 
  function)
  codec_zap.c:220: error: ?ZT_TRANSCODE_OP? undeclared (first use in 
 this 
  function)
  codec_zap.c:223: error: dereferencing pointer to incomplete type
  codec_zap.c: In function ?zap_new_alawtog723?:
  codec_zap.c:240: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in 
 this 
  function)
  codec_zap.c:262: error: dereferencing pointer to incomplete type
  codec_zap.c:269: error: dereferencing pointer to incomplete type
  codec_zap.c:269: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in 
  this function)
  codec_zap.c:270: error: dereferencing pointer to incomplete type
  codec_zap.c:271: error: dereferencing pointer to incomplete type
  codec_zap.c:277: error: dereferencing pointer to incomplete type
  codec_zap.c:278: error: dereferencing pointer to incomplete type
  codec_zap.c:279: error: ?ZT_TRANSCODE_OP? undeclared (first use in 
 this 
  function)
  codec_zap.c:281: error: dereferencing pointer to incomplete type
  codec_zap.c: In function ?zap_new_ulawtog723?:
  codec_zap.c:297: error: ?ZT_TCOP_ALLOCATE? undeclared (first use in 
 this 
  function)
  codec_zap.c:319: error: dereferencing pointer to incomplete type
  codec_zap.c:326: error: dereferencing pointer to incomplete type
  codec_zap.c:326: error: ?ZT_TRANSCODE_MAGIC? undeclared (first use in 
  this function)
  codec_zap.c:327: error: dereferencing pointer to incomplete type
  codec_zap.c

Re: [asterisk-users] got-name

[Daryl Jones]

Bill Michaelson wrote:
 Is it just me, or is the AGI interface at cnam.got-name.com failing 
 for others? Anyone know how to contact them without sending postal 
 mail or telegram?

I don't know how to contact them, but I am having the same problem.



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Re: [asterisk-users] Cisco remote reboot

[Daryl Jurbala]

I've fired a script from an AGI-BIN to accomplish that.

Try this one:

#!/usr/bin/perl
# mk 2004  feel free to distribute
#  [EMAIL PROTECTED], _Vile
#  perl script to reboot phones
#  try telnetting to your phone, first.
#
use Net::Telnet ();

$phone_ip = shift;

# Your Cisco 79xx prompt
$prompt = Enter Your Prompt Here;

# Your Password
$password = xx;

# Reset Command
$command = reset;

if ($phone_ip eq all)
{
reboot(xxx.xx.x.xx,$password,$command,$prompt);
reboot(xxx.xx.x.xx,$password,$command,$prompt);
reboot(xxx.xx.x.xx,$password,$command,$prompt);
reboot(xxx.xx.x.xx,$password,$command,$prompt);
} elsif ($phone_ip eq ) {
print Enter an IP or 'all' for All.;
} else {
reboot($phone_ip,$password,$command,$prompt);
}

exit;

sub reboot{

my ($ip,$password,$command,$prompt) = @_;

$t = new Net::Telnet;
$t-open($ip);

$t-waitfor('/Password :.*$/');
$t-print($password);

$t-waitfor('/'.$prompt.'.*$/');
$t-print($command);

}


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Re: [asterisk-users] Asterisk High-Capacity Stability

[Daryl Jurbala]

On May 14, 2007, at 11:27 PM, Atlanticnynex wrote:

I'm curious what kind of configuration/features/modules you could  
recommend for my setup. Can you explain further what you mean by  
OpenSER to Asterisk?


If you want to go Open Source, I think OpenSER is a good choice.  You  
won't need to do any hacking to make it work..I'd suggest making  
1 or 2 openser boxes to act as registrars for your user agents, and  
use the openser dispatcher module to point at one or more openser  
boxes that do LCR for calls that go directly out, and at one or more  
asterisk boxes for feature servers if you need them.


Using Asterisk realtime and the database extensions for OpenSER you  
can share the user database between them and things should just  
work.  Write your CDRs to a separate database (as to separate  
business data and call flow datajust in case someone does a  
complex CDR query you don't want your PDD to go through the roof) and  
come up with some kind of CDR remediataion for billing.


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Re: [asterisk-users] Asterisk High-Capacity Stability

[Daryl Jurbala]

On May 12, 2007, at 4:11 PM, Atlanticnynex wrote:


Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since  
I have quite a bit of experience there, and very little with SER.  
At this point, I'm wondering from a dimensioning standpoint, what  
kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB  
RAM). As I said, I don't plan to do any transcoding. I read the  
voip-info page on dimensioning and it seems theres some mixed  
feelings about Asterisk in high-capacity environments. I guess I'm  
looking for input as to whether Asterisk could handle roughly one  
DS3's worth of calls (672 calls) just doing the LCR (I've seen some  
pre-built LCR apps, looks like they all do on-the-fly MySQL  
queries- I think I'd write my own AGI that would use a cache).



With my hardware, could Asterisk run stable for this amount of  
traffic?

What stability issues does Asterisk have at this scale?



Simply put, NO.  I am on a project now where a client had an OpenSER  
box acting as an SBC and registrar passing traffic to several  
asterisk boxes which are doing LCR lookups on the fly as well as  
writing custom CDRs all through PHP AGI scripts to a Postgres DB.   
The Asterisk boxes do not scale, and randomly start swallowing calls  
or, more often, restart the process (safe_asterisk is handling  
this).  There is some light IVR type usage for reporting account  
balances and the like.  With anything more than 80 or 90 calls on the  
box, the IVR prompts start to break up.  Ben through replacing  
hardware, more memory, different Asterisk builds, etc.


I've had an open issue with Digium support on this for at least a  
couple of weeks, and the best advice so far was try using the SVN  
build.  That makes things better, but it's still not anywhere close  
to fixed..


It's absolutely incredible that Asterisk works at all for some of the  
situations its been put in - major kudos to the developers.  But I  
don't think using it for what you're talking about is a long-term  
business strategy.  When the highlight of the 1.6 release is bridging  
channels, you know high volume sip to sip usage in a carrier class  
call routing environment is NOT what development is focused on.  And  
that's fine.  If you use a wrench to do the job of a screwdriver, you  
shouldn't complain when you bust your knuckles


That being said, I don't meant to trash Asterisk at all.  It's a  
fantastic feature server, and a great PBX, both of which things I use  
it for very successfully.  I just don't think it's ready to handle  
50k plus minutes a day SIP to SIP with LCR and billing data, no  
matter what you do with it.  I'm 100% positive there are people out  
there doing it successfully, but those are the exception, not the  
rule.  And I doubt they are running unmodified code.


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Re: [asterisk-users] Asterisk High-Capacity Stability

[Daryl Jurbala]

On May 14, 2007, at 1:29 PM, Zoa wrote:



Several people do use it for handling  50k minutes a day. (I'm one  
of them).
Yes, you need to know what you are doing, and have a nice design,  
but it is possible.Our code is only slightly altered. (mainly for  
billing purposes).


That's great if you're good enough/have the time to make that  
happen.  But when I have issues and call/pay Digium and don't get  
timely or meaningful answers, it's doesn't make for a good business  
decision to continue using it for that purpose when I can toss in a  
Nextone or Sansay and have it just work.  All the time.  No  
babysitting.  Full professional and timely problem resolution from  
the vendor, etc, etc, etc.  Don't even get me started on Digium not  
being able to get TC400Bs to properly negotiate g.723.1 5.3k when a  
client requests 6.3k first (thank god for Cantata).


I guess it all comes down to whether you want things to just work and  
be able to have tier 1/2 support capable of actually doing anything  
meaningful, or if you want to have the engineering level people  
forced to do all the work.  From my standpoint, the smart business  
decision is quite clear.


But, as I said, Asterisk is still driving the feature servers, and  
works well for it.  As mentioned by someone else previously in the  
thread, it makes a great endpoint.


If you're having good success with it, that's fantastic.  I would  
hope that you contribute back to the list how you set things up to  
make this a possibility.


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[asterisk-users] Packet2Packet Bridging Questions

[Daryl Jurbala]

I'm just starting to upgrade some boxes from 1.2.x to 1.4.1 as well  
as trying to get some of the RTP traffic offloaded from the network.   
I think I'm misunderstanding what the console messages mean when it  
says Packet2Packet Bridding SIP/blah to SIP/blah.  I though that  
meant that it had successfully (re)INVITED and the media was no  
longer going through my Asterisk box, but ethereal says different.


I'm not having much luck finding any information on this on the wiki  
or google.  Can someone point me in the right direction?


Thanks,
Daryl



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Re: [asterisk-users] Packet2Packet Bridging Questions

[Daryl Jurbala]

OK...that makes much more sense.  So here's my follow-up question:  
what's the easiest way to check if I'm native bridging a call.  I'm  
trying to offload as much RTP traffic as possible, and want to have a  
way to check quickly (there are well over 50 calls on each of these  
boxes at any given time).  I've been going the ethereal route, which  
is great for debugging, but not so good for a quick look.


Thanks again,
Daryl

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[asterisk-users] 2 ring delay before asterisk answer

[Daryl Sayers]

I am a little green when it comes to all this but I am trying to connect
our PBX to an asterisk server using a TDM400 with 4 FXO modules. I am able
to dial an extension on my PBX handset and I get a dialtone from the PBX.
After 2 rings I then hear the asterisk server connect and I get a dialtone
from asterisk. I am then able to dial an extension on another asterisk
server.

My question is: How do I get asterisk to connect immediately without the
annoying 2 ring wait before I can start dialing a number.

snippets of extensions.conf

[net_incoming]
exten = s,1,DISA(no-password,net_outgoing)


[net_outgoing]
exten = _2XXX,1,Dial(${PYRMONT}/${EXTEN:1})
exten = _2XXX,n,Hangup()

logging:
Jan 23 07:39:47 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring 
Begin)...
Jan 23 07:39:49 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 2 
(Ring/Answered)...
Jan 23 07:39:50 NOTICE[16526]: chan_zap.c:6184 ss_thread: Got event 18 (Ring 
Begin)...
-- Executing DISA(Zap/1-1, no-password|net_outgoing) in new stack

-- 
Daryl Sayers Direct: +612 95525510
Corinthian Engineering   Office: +612 95525500
Suite 54, Jones Bay Wharf   Fax: +612 95525549
26-32 Pirrama Rd  email: [EMAIL PROTECTED]
Pyrmont NSW 2009 Australia  www: http://www.ci.com.au
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[asterisk-users] CallerID number not being displayed on SIP phones

[Daryl Jones]

I'm having trouble with Cisco 7960 and Linksys SPA-942 SIP phones not 
displaying the Caller-ID number.  The Caller-ID name is displayed, but 
not the number.  Instead, the phones always display the value that's set 
in the fromuser= parameter in sip.conf.  If fromuser= is not set, then 
the literal asterisk is displayed in the calling number field on the 
telephone sets.


Can I dynamically set the fromuser= value to the CallerID number in 
extensions.conf?


How can I solve this problem?

Asterisk v1.2.9
Cisco 7960 firmware P0S-3-07-4-00.


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Re: [asterisk-users] Re: Rewriting caller ID from database?

[Daryl Jones]

Steven wrote:

There are two I can think of.
Hoodahek and asterdex (or asteridex)

We used hoodahek at first, but now use asterdex(sp?)
It has a web interface to enter the new names into.

We use it to fixup, corp. cell phones and used to use it for our leagcy PBX 
extensions.
  


I use some custom scripts to do database lookups and rewrite CallerID 
information.  Everything works fine with regard to the CID name, however 
my Cisco 7960 and Linksys SPA-942 phones do not display the calling 
number. Instead, they display the called number.  This makes the phone's 
call return feature not work. The calling number and name are both 
properly displayed on all of the softphone clients that I've tried.


Here's the format I'm using to set the CallerID.

   SET CALLERID JONES DARYL A6508701826


Can anyone help?


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Re: [asterisk-users] Cisco 7960 SIP 8-3-0

[Daryl Johnson]

Tim,

I have seen the same 400 errors and the broken MWI...  I backed up to 
7.3...  We'll see if Cisco corrects these in the next release...


Daryl

- Original Message - 
From: Tim Connolly [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Monday, July 17, 2006 12:06 PM
Subject: [asterisk-users] Cisco 7960 SIP 8-3-0



Looks like the MWI broke on 8-3 also...
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[Asterisk-Users] Need help with two-stage ringing macro

[Daryl Jones]

I've been using the following macro to ring SIP and IAX devices for a 
few seconds, and then add on a cell phone if there is no answer on the 
SIP or IAX device.  Periodic problems began a few versions ago and now 
the problem happens every time with 1.2.9 and 1.2.9.1. 

The problem is that when a call from the PRI falls through to voicemail, 
the call is dropped before the voicemail greeting is heard.  Debug shows 
that voicemail is starting and that Asterisk is dropping the call on the 
PRI.   Calls made from SIP or IAX devices work fine.



[macro-followme]
;
; modified standard extension macro for two-stage ringing.
;
;  It will call the destinations in ${ARG4} for ${ARG2} seconds, and
;  if that fails, the destinations in ${ARG5} for ${ARG3} seconds.  If
;  that also fails, it will send the call to voice mail for extension
;  ${ARG1}.
;
;  Note:  if you want it to ring phone1 first, then phone1 AND phone2
;  next, you have to list phone1 in both lists.  Otherwise it will
;  stop ringing on phone1.
;
;   ${ARG1} - voice mail context
;   ${ARG2} - Extension
;   ${ARG3} - Time to ring stage 1
;   ${ARG4} - Time to ring state 1 + 2
;   ${ARG5} - Device(s) to ring stage 1
;   ${ARG6} - Device(s) to ring stage 2
;
exten = s,1,SetCallerID(${CALLERIDNUM:-10:10}) ; Send only the last 10 
digits

exten = s,2,NoOp(CallerID After:${CALLERIDNUM})
exten = s,3,SetAccount(${ARG2})
exten = s,4,Dial(${ARG5},${ARG3},rt)   ; Ring the primary group
exten = s,5,Dial(${ARG5}${ARG6},${ARG4},rt)  ; Add in the secondary group
exten = s,6,Voicemail([EMAIL PROTECTED]) ; send to vm as unavail
exten = s,7,Hangup
exten = s,106,Voicemail([EMAIL PROTECTED]) ; send to vm w/ busy 
announce

exten = s,107,Hangup

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Re: [Asterisk-Users] Re: Asterisk and 7960s

[Daryl Jones]

The @ip-address is actually a documented cisco fix to another problem.
I'd have to look it up, cause I don't remember exactly what it was, but
it's been on the list somewhere, and I think EVERYONE that's used 8.2 has
the same problem with the firmware.  I would suggest using 7.4 or 7.5.



I've been following this thread and it's clear is mud...  Would someone 
care to summarize?


Is it possible to automatically display the caller's number (true ANI in 
my case), caller name and caller address on a 7960 that's running 8.2?  
We currently rewrite CID Number and CID Name with a PHP script that does 
database lookups, however we can't get anything more than the name to 
display on the 7960.


If this is possible, we sure would appreciate a summary of how you did it.



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[Asterisk-Users] Problems with incoming PSTN calls

[Daryl St Amand]

I am having problems getting incoming 
calls from the PSTN to route to extensions, digital receptionist and even 
voicemail. When I call a DID number for one of the lines, it rings twice 
then says: "Goodbye" and hangs up. (logs to follow below configuration info). 
When I dial  it goes to the digital receptionist without any 
problems. The system setup is simple; I have 8 PSTN lines 
incoming to an Adtran 750, which is then connected to a TE110P via a crossover 
T-1 cable. I am running Asterisk 1.2.1 and AMP to help with basic 
configuration. Inside AMP, I have the inbound routing set to redirect to 
the digital receptionist. I have 4 SoundPoint 501 SIP phones and a 
SoundPoint 601 SIP phone. I can successfully call between extensions and 
make outbound calls. The "full" log for Asterisk shows the following: 
Jan 19 15:17:57 VERBOSE[24046] logger.c: -- Starting simple switch on 
'Zap/1-1' Jan 19 15:18:01 NOTICE[24046] chan_zap.c: Got event 18 (Ring 
Begin)... Jan 19 15:18:03 NOTICE[24046] chan_zap.c: Got event 2 
(Ring/Answered)... Jan 19 15:18:03 VERBOSE[24046] logger.c: -- Executing 
Playback("Zap/1-1", "vm-goodbye") in new stack Jan 19 15:18:03 DEBUG[24046] 
chan_zap.c: Took Zap/1-1 off hook Jan 19 15:18:03 DEBUG[24046] chan_zap.c: 
Enabled echo cancellation on channel 1 Jan 19 15:18:03 DEBUG[24046] 
chan_zap.c: No echo training requested Jan 19 15:18:03 DEBUG[24046] 
channel.c: Scheduling timer at 160 sample intervals Jan 19 15:18:03 
VERBOSE[24046] logger.c: -- Playing 'vm-goodbye' (language 'en') Jan 19 
15:18:04 DEBUG[24046] channel.c: Scheduling timer at 0 sample intervals Jan 
19 15:18:04 DEBUG[24046] channel.c: Scheduling timer at 0 sample intervals 
Jan 19 15:18:04 VERBOSE[24046] logger.c: -- Executing Macro("Zap/1-1", 
"hangupcall") in new stack Jan 19 15:18:04 VERBOSE[24046] logger.c: -- 
Executing ResetCDR("Zap/1-1", "w") in new stack Jan 19 15:18:04 DEBUG[24046] 
pbx.c: Function result is '(null)' Jan 19 15:18:04 DEBUG[24046] pbx.c: 
Function result is '(null)' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function 
result is 's' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'default' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 'Zap/1-1' 
Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is '(null)' Jan 19 
15:18:04 DEBUG[24046] pbx.c: Function result is 'ResetCDR' Jan 19 15:18:04 
DEBUG[24046] pbx.c: Function result is 'w' Jan 19 15:18:04 DEBUG[24046] 
pbx.c: Function result is '2006-01-19 15:18:03' Jan 19 15:18:04 DEBUG[24046] 
pbx.c: Function result is '2006-01-19 15:18:03' Jan 19 15:18:04 DEBUG[24046] 
pbx.c: Function result is '2006-01-19 15:18:04' Jan 19 15:18:04 DEBUG[24046] 
pbx.c: Function result is '1' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function 
result is '1' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'ANSWERED' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'DOCUMENTATION' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'(null)' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'1137701877.38' Jan 19 15:18:04 DEBUG[24046] pbx.c: Function result is 
'(null)' Jan 19 15:18:04 VERBOSE[24046] logger.c: -- Executing 
NoCDR("Zap/1-1", "") in new stack Jan 19 15:18:04 WARNING[24046] cdr.c: CDR 
on channel 'Zap/1-1' not posted Jan 19 15:18:04 WARNING[24046] cdr.c: CDR on 
channel 'Zap/1-1' lacks end Jan 19 15:18:04 VERBOSE[24046] logger.c: -- 
Executing Wait("Zap/1-1", "5") in new stack Jan 19 15:18:05 DEBUG[24046] 
chan_zap.c: Exception on 28, channel 1 Jan 19 15:18:05 DEBUG[24046] 
chan_zap.c: Got event On hook(1) on channel 1 (index 0) Jan 19 15:18:05 
DEBUG[24046] chan_zap.c: disabled echo cancellation on channel 1 Jan 19 
15:18:05 VERBOSE[24046] logger.c: == Spawn extension (macro-hangupcall, s, 3) 
exited non-zero on 'Zap/1-1' in macro 'hangupcall' Jan 19 15:18:05 
VERBOSE[24046] logger.c: == Spawn extension (default, s, 2) exited non-zero on 
'Zap/1-1' Jan 19 15:18:05 DEBUG[24046] chan_zap.c: Hangup: channel: 1 index 
= 0, normal = 28, callwait = -1, thirdcall = -1 Jan 19 15:18:05 DEBUG[24046] 
chan_zap.c: disabled echo cancellation on channel 1 Jan 19 15:18:05 
DEBUG[24046] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1 Jan 
19 15:18:05 DEBUG[24046] chan_zap.c: Updated conferencing on 1, with 0 
conference users Jan 19 15:18:05 VERBOSE[24046] logger.c: -- Hungup 
'Zap/1-1' Thanks,

Daryl 

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Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ...error while writing audio data!

[Daryl Johnson]

I had similar problems...  I downloaded the OH323 package for [EMAIL PROTECTED] 
and installed it 
(https://sourceforge.net/project/showfiles.php?group_id=123387)...  Seems to 
work much better.  It includes:


   gnugk 2.2.1
   pwlib 1.6
   open h323 1.13

I have setup my Cisco uBR924 with H.323 and I can place outbound calls.  The 
only issue I have is sending inbound calls to the Cisco device.  Any 
thoughts???


Daryl

- Original Message - 
From: Adam Rybak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Sent: Sunday, November 27, 2005 4:21 PM
Subject: Re: [Asterisk-Users] OH323 channel in asterisk 1.2 ...Ouch ...error 
while writing audio data!



It looks like compiling oh323 with wrong version of headers or wrong 
version of

open323/pwlib.  Are you completly sure that you deleted old headers and
libraries when upgraded asterisk to new version?

Adam Rybak

Cytowanie Rafael R. GV [EMAIL PROTECTED]:


/var/log/asterisk/full.1 output:

Nov 26 21:25:39 VERBOSE[14215] logger.c:  [chan_oh323.so]Nov 26 21:25:39
WARNING[14215] loader.c: /usr/lib/asterisk/modules/chan_oh3
23.so: undefined symbol: _ZNK8PChannel7IsClassEPKc
Nov 26 21:25:39 WARNING[14215] loader.c: Loading module chan_oh323.so
failed!

thanks
rafael




On 11/27/05, Adam Rybak [EMAIL PROTECTED] wrote:

 You should have more info in full log messages, look to this file and 
 send

 output.

 Adam

 Cytowanie Rafael R. GV [EMAIL PROTECTED]:

  Hello
  I am trying to compile oh323, oh323-0.6.5 wont compile with asterisk
  1.2libraries, must be
  oh323-0.7.3, now I have compiled this version but when reload 
  asterisk i

  have this error:
 
  [chan_oh323.so]Ouch ... error while writing audio data: : Broken pipe
 
  Any idea???
 
  --
 
  rrgv
 



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--

rrgv




Pozdrawiam,
Adam Rybak
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[Asterisk-Users] OT: SIP firmware image for Cisco 7940 or 7960

[Daryl Johnson]

Sorry for the off topic message, but I am ready to give up on this 7940...

I don't know what firmware version is loaded, but based on the sniffer 
traces it appears to be SIP 5.x or better...  The problem is that I don't 
have any firmware files for this device.  Can anyone point me in the right 
direction?


Thanks for the help,
Daryl 


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Re: [Asterisk-Users] * 1.0.8: no more reacting to callerid?

[Daryl Jones]

It's not just you.  Same thing happens here. I went back to 1.0.7.

Stefan Gofferje wrote:

Hi folks,

I used to have some constructions like

exten = number/callerid,1,Goto(somewhere)

After updating to 1.0.8 those does not work any more.
Any hints?

Regards,
Stefan


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RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 John Cianfarani
 Sent: Tuesday, June 21, 2005 8:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] ee1000 Ethernet in Dell 1850
 
 Does anyone know what the reason why Dell servers cause so 
 many problems for the digium hardware?
 Better question any Dell models that don't have any these 
 problems with the digium hardware?

I've got a PowerEdge 1400SC (old, P4 1gHz, upgradable to dual proc)
that's been a absolute tank.  Got 2 TDM400P's in it and it supports a
very small office with mixed SIP and POTS inbound/outbound.  Running
Debian, of course.

Daryl
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RE: [Asterisk-Users] WiFi IP Phones

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Anton Krall
 Sent: Wednesday, June 15, 2005 3:01 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] WiFi IP Phones
 
 Guys.
 
 I know there are wifi sip phones out there but I have a 
 question, are any of these phones anti explosive? By that I 
 mean, there are certain regulations about phones or cel 
 phones that are not recommended to operate in environments 
 like gas stations due to sparks and the chance of ingiting gas fumes.

You are referring to (in the US anyway) certification as intrinsically
safe.

I don't know either way about phones listed as such, but with the right
terminology you might have better liuck searching.

 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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RE: [Asterisk-Users] WiFi IP Phones

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dean Collins
 Sent: Thursday, June 16, 2005 9:57 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] WiFi IP Phones
 
 Ahmm Andrew, are you sure they are steel?
 
 It's been a long time since I did any work in this space but 
 we used to install them in plastic not metal.plastic 
 works better with the radio waves.

IS does not necessarily mean steel.  My Motorola alpha pager, and my
Motorola XTS3000 radio are both plastic and IS listed.


 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Huddleston, Robert
 Sent: Tuesday, June 14, 2005 3:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
 
 Anyone paying over $450 for a T1 is being ripped off...
 If you are in VA,MD,DC,PA,DE,NJ you can get an integrated 
 VoIP T1 for $300 - $400 and a flat internet t1 for about $400.
 The integrated VoIP T1 is great because it's handed off as an 
 ethernet - no need for a csu/dsu 

Ummm...no.  Maybe if you are in or very near a city you can, but not
everywhere.

You find me a reliable Teir 1 ISP T1 in New Hope, PA for $300 to $400
and I'll give you the amount I save over the next quarter.  NPA-NXX is
215-862.  Good luck.

 voiceverified. | Daryl G. Jurbala
 -- | Chief Technology Officer
| 215.862.1160 x235 (Office)
It had to be you!   | 215.862.9880 (FAX) 


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RE: [Asterisk-Users] UPS rating for SOHO asterisk box

[Daryl G. Jurbala]

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Terry H. Gilsenan
 Sent: Wednesday, June 01, 2005 5:05 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] UPS rating for SOHO asterisk box
 
 
 I have many sites that have a 35amp Charger with 2 x 400ah 
 900CCA deep cycle batteries (10 year warranty), and 1000VA inverters.
 
 The combination makes for perfect power and about 2.5 days 
 run time with my network kit whish consists of several Dlink 
 wifi access points, 1 xbox (hacked into a router/firewall) 
 and a vsat system.
 
 Total cost for the power kit AUD$1400 all up, and not a 
 single second of downtime in over a year.
[...]

Yepyou can (somewhat) build your own UPS with peoperly rated
equipment.  As a matter of fact, most telco installations don't have
monolithic UPS's (like you'll see in most larger datacentersyou
know..the big box that says Liebert on it), they use racks of batteries
with separate charging circuits.  Most of the equipment runs directly
off of the battery voltage, but you will find places with some inverters
as well.  Of course, the room is properly designed (spaced,
non-combustible racks, fire detection and supression systems, etc.) and,
in most jurisdictions they also have to carry one or more operational
permits (current Internation Fire Code requires permitting for stationar
lead-acid battery systems exceeding 50 gallons liquid capacity). 

 On the flipside, I have seen a ups flare when the transformer 
 overheated and melted the varnish, nasty!

I've seen completely unmodified (although not properly maintained) UPSes
as large as 5000 Va completely melt down to the point where they
destroyed their own chassis, damaged the rack they were sitting in, and
activated the clean-agent supression system in the rooms they were in.
This was actually a big problem with one of my customersthey hadn't
been maintaining their UPSesthe replace battery lights had been
lit for months (they had all been purchased at about the same time).
Within a span of about 3 months, 4 of them melted down similarly.  A
quick call to APC revealed that the batteries in these units were rated
for about 12 monts less than they had actually been in service, and a
simple battery replacement would have prevented the problem (the chassis
was rated for something like 3 sets of batteries...whatever the lifespan
of the batteries was3 years I believe).

So, don't do stupid things with high voltage, like modifying equipment
that wasn't meant to be modified, using undersized equipment, failing to
properly vent batteries, or storing your contraption on or near
combustibles.  It's just NOT worth the risk.  Take it from someone who's
pulled his share of bodies (of both the live and dead varities) out of
buildings.  I've seen way too many fires started by electrical system or
device modifications similar to those described in previous posts.
And most people who do things like this just never consider the life
safety risk involved until its way too late.

I'll get off my soap-box now and get back on topic.

Daryl
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RE: [Asterisk-Users] UPS rating for SOHO asterisk box

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jean-Michel Hiver
 Sent: Tuesday, May 31, 2005 5:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box
 
[...]
 Regarding this, I have done this hack yesterday:
 
 - Remove the battery from an existing UPS
 - Rewire the UPS onto biggest car lead acid battery (12v) you 
 can find.
 
 Et voila! Bigger capacity. Put the batteries in parrallel and 
 you do get monstruous UPS capacity... the only trouble with 
 it is that re-charging the batteries may take some time.
[...]

Congratulationsyou've just given this part-time small town fire
marshal and 14-year fire service veteran nightmares.

Kidsdo NOT try this at home.  The inverters in small UPSes are not
designed to deal with runtimes that exceed the batteries in them.  If
you run this setup well past the time it was designed to run (by adding
3, 4, or more times that battery capacity it was ever designed to have)
that chances of a catastrophic inverter failure (meaning flash, boom,
fire) are very real and very likely.

Daryl
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RE: [Asterisk-Users] Did nufone change allowed codecs?

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Wilson Pickett
 Sent: Thursday, May 05, 2005 7:11 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Did nufone change allowed codecs?
 
 Hi,
 
 I've been using nufone DIDs for months with no problem. Now 
[...]
 No files will play on a call to asterisk because they aren't 
 found in g729. Perhaps the desired codecs for DID have 
 changed? I know you can specify them when ordering DID, but I 
 see no way to change them once the DID are provisioned.
 
 Any help?

Just a crazy idea herehave you contacted NuFone support yet?

Daryl
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RE: [Asterisk-Users] 1800 DNIS and asterisk (HOW TO?)

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Thursday, May 05, 2005 6:20 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] 1800 DNIS and asterisk (HOW TO?)
 
 Hi
 I have applied for Qwest 1800 termination to a T1 ISDN PRI. 
 They tell me that I will have to program a predefined DNIS 
 number on my switch. 
 According to them unless asterisk returns that DNIS number no 
 call will get through.
 
 How do I program the DNIS, is it through zaptel.conf or some 
 other way. Is it required??. As per qwest if the 8xx # is 
 going to be routing to an ISDN TG, DNIS is required.
[...]

Huh?  The last time I dealt with DNIS (admittedly, years ago) the
provider sent the digits to ME via DTMF to tell ME what number was
dialed to terminate on that line (you knowDialed Number
Identification Service).

Unless DNIS has turned into telcoBGP while I haven't been watching, what
you're being asked to do doesn't seem quite right.

Daryl
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[Asterisk-Users] Asterisk on VMWare ESX/blade servers

[Daryl G. Jurbala]

Has anyone had any experience (good or bad) running Asterisk under
VMWare ESX server on a blade chassis?  This application will (fairly
obviously) not include Zap channelsactually, it will be SIP-only.

Please feel free to contact me off-list and I'll summarize for the list
later.

Daryl G. Jurbala
NGM Tec, Inc.
Tel: 215-862-1160 ext. 235
Fax: 215-862-9880 
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RE: [Asterisk-Users] small qos switch

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Andrew Latham
 Sent: Sunday, March 27, 2005 12:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] small qos switch
 
 I heard a great solution at Linux World Boston. A rather 
 talented young man mentioned using a IPV6 VPN on the IPV4 
 internet. IPV6 supports QOS by default. Just VPN straight 
 back to the CO and have your POP there so you only need one 
 firewall too.

He may have been talented, just not in network engineering.

While your IPv6 encapsulated VPN would have QOS, the underlying
transport medium (IPv4) still would not (if it didn't have it before).
Furthermore, if any Ipv4 hops in between would have prioritized your
traffic higher based on its type, they now have no idea what is is,
because it's encapsulated.

Daryl
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RE: [Asterisk-Users] How NuFone.Net's customer service works.

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Linn Boyd
 Sent: Monday, March 14, 2005 6:05 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] How NuFone.Net's customer service works.
 
 Hello All,
 
 I have been using asterisk for some time, and I would 
 like for all to take a look at what NuFone does when they get 
[...]
 I hope that people 
 that care about customer service avoid NuFone.net
[...]

Hmmm...I've had 2 problem with my NuFone service in the year or more
I've used them.  Each time I've treated them professionally when
reporting the issue and received the same treatment in return.  The
issues were also resolved promptly.

Daryl
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RE: [Asterisk-Users] Which hardware for this solution?

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Giorgio Mandolfo
 Sent: Wednesday, March 09, 2005 8:59 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Which hardware for this solution?
 
 Hello,
 
 we are a firm who wants to develop some VOIP solutions.
[...]

 Straight to the point: what kind of hardware I need? I saw 
 some PCI cards (like Digium Wildcard TE110P) but I am not 
 sure what to buy.

You need to but the appropriate cards to interface with the PBX you are
trying to connect to.  Without knowing what interfaces it has available,
that's a difficult question to answer.

If it's got an E1 or T1 interface, buy an appropriate port-density T1/E1
card (surprise) like a TE110P or TE410/405P.  If it's analog, and
appropritaely-configured TDM400P would be the way to go.

Cards are cardsget what you need to make the interface happen.  It's
like asking what card you need to connect your computer to some
undescribed network.  If the network is ethernet, you need an ethernet
card.  If it's token ring, you need a token ring card, etc.
Daryl
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[Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice

[Daryl G. Jurbala]

I'm looking for an application that can monitor a channel for voice
input and then proceed on.  The closest thing I've found is
BackgroundDetect, which expects DTMF.

Here's what I'm doing:
-Call file generated which calls someone and connects them to an
extension.
-Extension plays stuff, etc. etc. etc (not important)

With digital or VoIP termination, this works fine, because * knows when
the line is answered.  On analog POTS, it has no idea when the call is
actually answered, only when its dialed, so the playback starts right
after the line is dialed, not after the called party picks up.

The Dialogic IVR SDK monitors call termination status this way, so I'm
looking for something similar in *.  Anyone have any ideas on this one?
Or am I going about this the hard way and missing an obvious
alternative?

Thanks,
Daryl
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RE: [Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice

[Daryl G. Jurbala]

Yes, I'm replying to my own post.

Roger Gulbranson suggested this:
http://www.voip-info.org/tiki-index.php?page=NVBackgroundDetect

As he's using it for FAX detect, and it has a talk option as well.

If anyone is interested, I'll report back with my results.

Thanks Roger!
Daryl
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RE: [Asterisk-Users] Detect sound and continue, like BackgroundDetect() for voice

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steve Underwood
 Sent: Thursday, March 03, 2005 10:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Detect sound and continue,like 
 BackgroundDetect() for voice
 
 Yes, you missed an obvious option - search the mailing list. 
 This has come up an number of times.

I searched both the Wiki, and the list.  But I obviously didn't come up
with the right search terms, or overlooked relevant results, which is
why I asked.

How about this: the next time I have a question to ask, I'll call you
first to ask for what search terms to use before posting.

What's your mobile number?

Thanks,
Daryl
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RE: [Asterisk-Users] RE:Terrible inbound call quality vs. outbound

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Gene Willingham
 Sent: Tuesday, February 01, 2005 6:49 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] RE:Terrible inbound call quality 
 vs. outbound 
 
 
 
 I am experiencing the same problem, except I do not use 
 Voicepulse outbound.
 I have 100 Mbps connection, so it should not be a bandwidth 
 issue.   Last
 Thursday they had a 4 hour outage on inbound calls.  The call 
 quality has deteriorated since.  I am in the process of 
 looking for another provider.
[...]

Not to just me too, butme too.  I've contacted their support on
numerous occasions, and have been given busywork to do (run ping plotter
for 24 hours, send us the results, etc) and never receive a response
that acknowledges a problem of any sort.

Daryl
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RE: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
 Sent: Tuesday, February 01, 2005 11:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Zap channel occasionally misses 
 dialing thefirst digit
 
 have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels
 
 Here it tells you that you can specify a wait period.
[...]

Don't know if it will apply to those having issues with BRI/PRI, but in
my case, a ww in front of the dial string has worked witout fail for the
last few days.

Thanks to all who helped,
Daryl
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[Asterisk-Users] FIX YOUR AUTO-RESPONDERS!!!

[Daryl G. Jurbala]

PLEASE CONFIGURE YOUR 
AUTORESPONSERS TO NOT SEND MESSAGES TO PEOPLE WHO POSTIN MAILING LISTS YOU 
SUBSCRIBE TO.

This is an extremely rude 
thing to allow, and is becoming increasingly common, especially with users of 
the Asterisk-Users list.

Daryl

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] Sent: Friday, February 04, 
  2005 6:41 PMTo: Daryl G. JurbalaSubject: AUTOREPLY RE: 
  [Asterisk-Users] Zap channel occasionally misses di...
  
  Vielen Dank für Ihre Email!
  
  Ich bin vom 02.02.05 bis einschließlich 13.02.05 ausser Haus.
  
  Ihre Email wird bis dahin nicht bearbeitet oder weitergeleitet.
  
  Bei dringenden Fragen wenden Sie sich bitte an meinen Kollegen
  
  Herrn Rüdiger Hoog
  Email: [EMAIL PROTECTED]
  Telefon: 02331/473101-11Telefax: 02331/473101-19
  
  
  Für weitere Fragen stehe ich Ihnen gerne zur Verfügung und 
  verbleibe
  
  mit freundlichem Gruß,
  
  Stefan SpeckenheuerTechnische Leitung
  
  POS Service, Logistik  Handels GmbHAuf dem Graskamp 
  2D-58099 Hagen
  
  Tel. +49 2331 473101-21FAX +49 2331 473101-39
  
  mailto:[EMAIL PROTECTED]http://www.posservice.de
  
  Sitz der Gesellschaft: Walter-Rathenau-Ring 9-11, 59581 Warstein 
  BeleckeHandelsregister Arnsberg: HRB 2958Ust.IdNr.: DE 198 933 
  818Geschäftsführer: Martin Menzel, Christian Woelke
  
  Diese E-Mail einschließlich aller Anhänge ist vertraulich.Wir 
  bitten, eine fehlgeleitete Mail unverzüglich vollständigzu löschen und uns 
  eine Nachricht zukommen zu lassen.Wir haben die Mail beim Ausgang auf 
  Viren geprüft;wir raten jedoch, auf Grund der Gefahr auf 
  denÜbertragungswegen, zu einer Eingangskontrolle.Eine Haftung für 
  Virenfreiheit schließen wir aus.
  
  This e-mail and any attachments are confidential.If you are not the 
  intended recipient of this e-mail,please immediately delete its contents 
  and notify us.This e-mail was checked for virus contamination 
  beforebeing sent; nevertheless, it is advisable to checkfor any 
  contamination occuring during transmission.We cannot accept any liability 
  for virus contamination.
  > -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Tuesday, February 01, 2005 11:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Zap channel occasionally misses 
> dialing thefirst digit
> 
> have a look at http://www.voip-info.org/wiki-Asterisk+zap+channels
> 
> Here it tells you that you can specify a wait period.
[...]

Don't know if it will apply to those having issues with BRI/PRI, but in
my case, a ww in front of the dial string has worked witout fail for the
last few days.

Thanks to all who helped,
Daryl
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[Asterisk-Users] Zap channel occasionally misses dialing the first digit

[Daryl G. Jurbala]

I THINK.  When dialing 1+10 digits, I occasionally get a telco
message You must first dial a 1.  When I look at the console, the
number is being sent to the ZAP channel properly.  We're talking about a
couple of POTS lines on a TDM400P.

I'm thinking that it may be starting the dial too early after coming
off-hook because I can just redial and have it work (or not) randomly.
Does anyone know what this might be and/or an easy way to have the ZAP
channel come off-hook, delay for 1/2 second or so, and then dial?

Thanks,
Daryl G. Jurbala
NGM Tec, Inc.
Tel: 215-862-1160 ext. 235
Fax: 215-862-9880 
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RE: [Asterisk-Users] Qs about FXO/FXS cards

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steven Critchfield
 Sent: Monday, January 03, 2005 4:55 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Qs about FXO/FXS cards
 
[...]
 For business use, I would suggest you first find a BRI card 
 you can use here in the states. Hint, bug Kapejod into making 
 that 4 port card US ready. Then move any business user over 
[...]

That might work out where you do your deployments.  In Verizon
territory, you can get analog business lines with unlimited long
distance and no metered minutes for about $37 a month.  A BRI costs you
about double that for the loop, with metered minutes and bring your own
LD.

Past the technology aspects, BRI just doesn't work here.  And I'm going
to guess that pricing structure is similar in other areas as well.
Daryl
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RE: [Asterisk-Users] Qs about FXO/FXS cards

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Saturday, January 01, 2005 9:12 PM
 To: asterisk-users@lists.digium.com
 Subject: RE: [Asterisk-Users] Qs about FXO/FXS cards
 
[...]
 I'm running older, but solid hardware and not seeing any 
 issues.  I'm using a Compaq Proliant 1850R Gen1 dual PII 400 
 with 512MB ram, GB ethernet, and SATA Hardware RAID.  Cheap, 
 efficient, redundant.  And for a Debian box, good enough.  
[...]

I just have to add my $0.02 here.  I've got a PIII-550 Proliant 800 that
NEVER has any issues like this.  It's running Debian woody, and has a
TDM400P that never has any of these issues.  It's also running 208v from
a high quality UPS.

As a telephone system should, it simply works.  It is forgotten about,
and used andused and used.  No one has to do much of anything to it, and
no one has to make excuses for it (sorry..it's VoIP).

Anyone who wants to run junk hardware and beta code pretty much loses
their right to complain about the results of doing so.

Daryl
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RE: [Asterisk-Users] Is H323 dying?

[Daryl G. Jurbala]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Martin List-Petersen
 Sent: Thursday, November 18, 2004 10:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Is H323 dying?
 
[...]
 That is correct. H.323 is something nobody real will deal 
 with, but it's still supported because a lot of the old 
 fashioned carriers do H.323.
[...]

Nobody real deals with it and it's supported by old fashioned carriers?

Please, don't thak this as an insult, but you need to qualify that your
background obviously doesn't include any carrier-class bulk VoIP
termination whatsoever when you make broad statement like that.

Millions and milions of minutes of voice and fax traffic each day are
carried over h.323, for end users that don't even know they are using
VoIP, and in most cases don't even know what VoIP is.  Minutes handled
by bold old and new companies.

Now if you wanted to say that it's not in vogue for soft PBXen and key
systems to support h.323, I'll buy that.  But I'm going to guess that
voice traffic over SIP is a mere fraction of voice traffic over h.323 on
any given day.

Daryl Jurbala
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Re: [Asterisk-Users] DTMF stops working w/ Voicemail

[Daryl Jones]

Brent Franks wrote:
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec.  This is all on an internal switched 100mb lan.
Has anyone else seen anything like this?
Confirmed...   Happens intermittently with Cisco 7960 phones for the 
past two weeks.  I haven't been able to identify what causes it to occur.

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RE: [Asterisk-Users] Asterisk and Linejacks

[daryl]

Nope...I scrapped that idea and just bought a Digium card. 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of greg
 Sent: Thursday, July 22, 2004 2:08 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Asterisk and Linejacks
 
 I found a message from you to the asterisk users mailing list 
 from 2001. I was wondering if you got (or still have) an 
 asterisk system working with the linejack? If so, would you 
 be willing to assist me with mine?
 
 I seem to have things working, and * says that caller ID is 
 coming in, but I can't get * to actually answer the call.
 
 Thanks,
 Greg
 
 --
 NetIO.org
 
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RE: [Asterisk-Users] 7960 Dynamic DNS?

[daryl]

 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Sunday, July 18, 2004 3:48 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] 7960 Dynamic DNS?
 
 I can't think of any router that supports this
 
 You could put it in as a request to www.sveasoft.com for 
 their firmware for the wrt54g (great box...runs linux and 
 lots of features and functionality).

Not only does the Sveasoft firmware already support dynamic DNS, the
original Linksys firmware does as well.  It was very common junk router
feature (and by junk I mean anything you can buy at Staples that claims
to be a router).
Daryl
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RE: [Asterisk-Users] Re: VoicePulse changes

[daryl]

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of 
 Randy Bush
  Sent: Thursday, July 15, 2004 3:35 PM
  To: [EMAIL PROTECTED]
  Cc: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Re: VoicePulse changes
  
  the message arrived here some hours after calls through 
 them stopped 
  working.  not very professional.  there should have been 
 considerable, 
  like multiple days, of overlap.  think about the customers 
 who are out 
  of reach of configuring their
  * server but still rely on the service.
 
 The message says AUGUST 15th.  This is JULY.  AUGUST 
 *follows* JULY concurrently/serially.

Yeah...that's believable.  Mike, wake upthey made changes, the broke
things.  While they obviously tried to give everyone a month's notice,
it just didn't work out that way.  While the old config still works
(now), I find it difficult to believe that the two events were not
related.  Especially when it automagically fixed itself.

Also like that I call in and get the automated status report, which
reports everythig fine, yet still wait on hold for over 45 minutes, and
find out that they actually already DO know that there is a problem.

None of it struck me as particularly professional.

If there is someone from Voicepulse here, feel free to stand up for
youself and tell us your side of things.  From here it's not looking too
good.

Daryl
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[Asterisk-Users] VoicePulse changes

[daryl]

I'm a bit displeased at the way this happened.  I received an email from
VoicePulse.  Here's some excerpts:

--
We're sending you this important update so you can take advantage of
improvements we've
been making to your VoicePulse Connect! service.

We've been working hard on improving the audio quality and reliability
of your Connect!
service, and this notice contains important information about
configuration changes you'll
need to make to maximize your performance.


[...]

DESCRIPTION OF CHANGE

New switches have been added to handle IAX2 TERMINATION.  The purpose
of this change
is to provide customers with the ability to use all of the latest audio
codecs available
in Asterisk's stable CVS branch, and to provide additional redundancy
for terminating calls.

Some key points regarding this change are:

- The new switches listed below are the latest stable branch of
Asterisk
- The previous method for terminating IAX2 calls using Connect! will
cease to be available
   at midnight (GMT) on August 15th, 2004.
- We recommend all customers using Connect! for IAX2 termination begin
using the new
   configuration immediately.

-

Note The previous method for terminating IAX2 calls using Connect! will
cease to be available at midnight (GMT) on August 15th, 2004.  The
message I got was at 1:51 AM EST.  That means I was given negative 5
hours and 51 minutes to make this change.

And even worse, my DIDs form them no longer work (sits as Request Sent
forevernever actually registers).

Just though I'd toss this one out there in case anyone else is having
problems, or possibly didn't see/didn't get the email.

Why, oh why is it so hard to find a stable DID in my area? (Philly)
IConnectHere is flaky, VoicePulsewell, we've covered that, and
NuFone, while rock solid and never giving me any problems, doesn't have
their non-Michigan DID's available yet.

And pleasemaybe I'm asking for too much, but any good DID provider
should be able to call forward your DID to whatever number you choose on
1.) failure of their systems or 2.) loss of connectivity to the remote,
no matter who's fault it is.  This should be automatic, seamless, and
the forwarding number should be changeable by the account holder on the
fly.

Daryl G. Jurbala
BMPC Network Operations
Tel (PA): +1 215 825 2107 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 215 862 9880
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp  
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RE: [Asterisk-Users] VoicePulse changes

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Andrew Yager
 Sent: Thursday, July 15, 2004 10:31 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] VoicePulse changes
 
 
  Note The previous method for terminating IAX2 calls using Connect! 
  will
  cease to be available at midnight (GMT) on August 15th, 2004.  The 
  message I got was at 1:51 AM EST.  That means I was given 
 negative 5 
  hours and 51 minutes to make this change.
 
 
 Check your clock. It's still July.

Whoopsstupid one on my part.  The combination of my neither
termination or origination working, plus this message made me simply
skim it.

Anyhoo, it looks like the old method isn't working at all anyway.
Switching to the new method for termination worksbut I still don't
have origination, and I've still waiting on hold (27 minutes and
counting).

Daryl
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RE: [Asterisk-Users] VoicePulse changes

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Rich Adamson
 Sent: Thursday, July 15, 2004 12:26 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] VoicePulse changes
 
[...]
 You should be far more disturbed with their comment about stable
 then with the rest of their email. Its a known fact that a 
 substantial number of 'fixes' have been made to Head and not 
 to Stable, and that's backed by a fair number of developers 
 including Mark.

Yes, yes...don't get me started.  But I'm sure you can understand how
not having working origination takes a bit of priority over their choice
of version.

FYI, origination magically started working aging about 45 minutes ago.

Daryl
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RE: [Asterisk-Users] VoicePulse changes

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk
 Sent: Thursday, July 15, 2004 1:00 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] VoicePulse changes
 
 Welcome back to July.  How's the future?
 
 There's one rather reliable, albeit not very popular provider 
 with DIDs in the Philly area:  Vonage.  Their softphone 

[...]

I, as well as almost everyone else on this list, is very well aware of
Vonage.  As soon as they start officially supporting Asterisk and
specify things like whether you can have concurrent inbound calls
without additional charge, calls rolling over, etc it just might be a
viable option.

No, my Asterisk installation is not in my basement being used as a
glorified answering machine.  People who use these things for actual
business systems need more than I played around with x and got y to
work!  Cool!  Maybe it will even keep working if x doesn't decide to
change it or start charging me, etc.

For now, that leaves people in my position paying for PRIs or POTS lines
just to be sure.

Daryl
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[Asterisk-Users] Intermittent cidname lookups

[Daryl Jones]

I'm having a problem with intermittent lookup of Caller ID Name info 
using LookupCIDName.

The same problem occurs when doing:
	asterisk -rx database show cidname
No data is returned on every fourth or fifth query. No errors are being 
logged.

I'm currently running CVS-HEAD-07/07/04-17:04:31 and first noticed the 
problem a few weeks ago.

Is anyone seeing a similar problem?
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Re: [Asterisk-Users] FINALLY! a good book about Asterisk.

[Daryl Jones]

I've been reading drafts of this book for at least nine months and can 
assure you that its content is very different than what is available on 
the Wiki. The book is an excellent introduction to VOIP in general, and 
offers sufficient information for the novice to configure a basic 
Asterisk system.  Advanced Asterisk users will still need the Wiki and 
mailing list archives.

Perhaps a future edition of the book might cover more advanced topics, 
but the first edition is intended for beginners.

Perhaps the most important thing that this book will accomplsih is to 
increase general awareness about Asterisk being a very reliable, 
full-featured PBX.


Harold Workman wrote:
what does that have to do with an overpriced book?
and i agree with Joe.  With this book sourcing most of the documentation
directly from wiki, why pay for something thats free?  Id rather donate $49
to keeping wiki free to the enviroment.
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[Asterisk-Users] Suggestions for 96 tip/ring lines?

[daryl]

Just starting to do the research on this oneI've got a customer who
is showing interest in replacing any older Panasonic unit providing
service to 96 tip/ring lines from a single PRI.  Does anyone have any
recent experience with a decent (as in, plays nice with * and has a
reasonable per-port cost) channel bank or similar?  

Mediatrix only goes up to 24 port, as far as I can tell, which puts me
around 13k of just their hardware.  And it just doesn't seem quite as
carrier class as a traditional channel bank to me.but I'm just going
on gut feeling hereplease feel free to correct me (nicely or not...I
really don't care).

Daryl G. Jurbala
BMPC Network Operations
Tel (PA): +1 215 825 2107 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 215 862 9880
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp  
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RE: [Asterisk-Users] Suggestions for 96 tip/ring lines?

[daryl]

Actually, if all of the outside lines are full than ca just get a
reorder tone for all it mattersbut yes, basically 96 desk stations
is what we're talking about.

Thanks for the pointer.  I'll look into the Adits.  Certainly sounds
like the price is right.

Daryl

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Andrew Kohlsmith
 Sent: Friday, July 02, 2004 9:24 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Suggestions for 96 tip/ring lines?
 
[...]

 Do you mean 96 desk phones with a PRI as the trunk interface 
 (i.e. a maximum 
 of 23 calls outside the building, but up to 96 phones in use)?
 
 A TE405P and a pair of Adit600s will give you 96 analog 
 channels.  I'm using a 
 T100P and a half-full Adit600 currently and it Just Works.
 
 TE405P is $1500, and a pair of 48-FXS-port Adit600s should 
 run you under $1200 
 or so, plus whatever you would pay for Amp D-50 to BIX for 
 the final wiring.
[...]
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RE: [Asterisk-Users] * and Cisco routers

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Lars Boegild Thomsen
 Sent: Tuesday, May 18, 2004 11:23 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] * and Cisco routers
[...]
 Speaking of which - anybody got 
 experience with VoIP and IPSec?  I've never really used 
 IPSec, but I would imagine it creates a significant delay.

I run one or more 7960's over several different VPN setups.  The one
that introduces the most latency is a cheap PIX (read: 501 or 506).  A
515 is OK, a 515 with a crypto card is pretty acceptable.  The best
setup is a 1721 or better with a crypto card.  I routinely run that
config at each end using GRE over IPSec and have no problems (it
introduces about 20 ms latency when properly configured.a cheap pix
can introduce about 40 to 80 on average).

One IPSec VPN connected between a 6509
MSFC-GigE-7206VXR-DS-3-7206VXR introduces only 12 ms latency on
average.  Of course that's nearly $30k worth of plumbing, so one would
expect that kind of performance.

Daryl
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RE: [Asterisk-Users] Cisco 7920 Image

[daryl]

And it actually is.the only problem is that the downloads on the
Cisco site are actually CallManager updates.  So you'd need a CM server
to extract the image file (which you could then toss on whatever tftp
server you want).

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian A.
Underwood
Sent: Thursday, May 06, 2004 11:52 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7920 Image


Trey Scarborough wrote:
 no you have to use a call manager server to upgrade the phone unless 
 you want to go through a larg ordeal to get the phones to upgrade 
 software. You will still have to have access to download callmanager 
 updates.

Like the 7960, I was expecting it to be as easy as specifying an OS in
the 
OS7290.txt file and having it TFTP over.  shrug

-- 
/* Ian A. Underwood - [EMAIL PROTECTED] - http://www.agentgreen.org
There are 4 boxes to use in the defense of liberty:
soap, ballot, jury, ammo. Use in that order. Starting now. */
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Re: [Asterisk-Users] WAMi - Windows Asterisk Manager

[Daryl Jones]

The default password is Admin.

Adam Goryachev wrote:

On Tue, 2004-04-06 at 14:36, Christian Hoffmeyer wrote:

Thank you for all of the beta testing.  New and improved graphics in this
release along with
drag and drop transfers and hold for all technologies.
There's a screenshot on the link below.  Also improved documentation so read
the included README.  There's also a sample xml configuration included.
http://www.voip-info.org/tiki-index.php?page=Asterisk+WAMI


I can't seem to find where I am supposed to create the config file, nor
do I know what the default admin password is.. 

Any suggestions?
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RE: [Asterisk-Users] avaya and linux

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Glen Ford
 Sent: Friday, April 02, 2004 2:48 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] avaya and linux
 
 
 Does anyone know if avaya voip product is running linux under 
 the hood?

Yes.  The 5300 (even the non-voip featured ones) are a RedHat
enterprise box with standard layer 2 switching hardware to connect the
chassis together.

Don't know about the other models, or even the current state of the 5300
platform, but the two or so year old ones I've been dealing with have
the above config.
Daryl
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Re: [Asterisk-Users] One voicemail - multiple boxes?

[Daryl Jones]

I contracted with Digium for this enhancement and am waiting for it to be 
completed.

Tilghman Lesher wrote:

On 2004 Apr 02, at 12:04, Brian Capouch wrote:

I don't want to re-invent the wheel if someone has already
hacked a way to do this.
One of my customers has a number of stores, and he wants
to leave one voicemail that would be delivered to all the
managers at once.  Each has a voicemail account on his server.
I have googled around and looked on the WIKI.  Maybe I'm
missing it?


There's a request to do this on the bugtracker, but no implementations
yet, AFAIK.

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RE: [Asterisk-Users] Cisco 7960 SIP Images

[daryl]

What you and so may others on this lise seem to forget is that Cisco is a company 
offering bsuiness products for businesses.  Businesses typically pay by check and wire 
transfer, especially for items such as this.
 
If you want home-user pay-by-credit-card service, buy products from Belkin's home line 
and similar.
 
Oh...what's that?  None of these cheesy Stocked-at-Costco hardware companies have any 
VoIP phones worth a crap?  Then deal with the fact that you are buying from a company 
who doesn't target home users, and deal with it.  It costs Cisco more money than they 
make on the contract to offer SmartNet on a single device like this.  You're lucky 
they don't have a minimum device limit/contract cost of something like 5 devices or 
$300/year.  I'm guessing this type of policy would hardly effect more than several 
hundred of their customers, most of them with 7960's and similar.

-Original Message- 
From: [EMAIL PROTECTED] on behalf of John Baker 
Sent: Sat 3/27/2004 4:41 PM 
To: [EMAIL PROTECTED] 
Cc: 
Subject: Re: [Asterisk-Users] Cisco 7960 SIP Images



[massive amounts trimmed]

No, you can't use a credit card.  You have to send the #$!@@$#'s a 
check.  It's really stupid, but it's the Cisco way. 

John 

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winmail.dat

RE: [Asterisk-Users] Home users

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Damian Dicks
 Sent: Sunday, March 21, 2004 9:18 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Home users
 
 
 I am trying to setup the following scenario.
[...]
 there is nothing.  I do have port forwarding turned on my 
 office firewall.
 
 Can someone help me here?  I am almost out of hair on my head.
[...]

canreinvite=no in sip.conf for the home phone.

Without that, the phoen are trying to talk to each other directly, which
isn't going to work when they are both (presumably) behind different NAT
boxes.  Canreinvite=no will force your home phone to always pass its
traffic through the * box, eliminating the issue you are having.

Daryl
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RE: [Asterisk-Users] 3com NBX phones

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Nicolas Bougues
 Sent: Friday, March 05, 2004 3:17 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] 3com NBX phones
 
 
[...]
 Note that the hardware is probably not the same as the 
 standard NBX phones : my SIP phones did feature an IR sensor 
 to be used by a Palm for automated dialing.

That's actually an option on the better NBX phones...your is probably a
2102-IR or similar, and has been since at least when I did my last NBX
rollout about a year and a half ago.  What seems different is that you
could flash it at all.  When connecting to an NBX, these phones grab
their firmware from the NBX they pin up to.  I suppose there is a
flashable area on the phone that is used as a boot loader in NBX mode,
and probably to store the whole image when flashed with SIP.

Can anyone confirm these are the same phones?  Because I still have
boxes of them somewhere too (that seems to be a common thread here).
Daryl
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RE: [Asterisk-Users] Dialing 800 numbers with VOIP

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Greg Boehnlein
 Sent: Monday, February 09, 2004 10:50 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Dialing 800 numbers with VOIP
 
 
 On Mon, 9 Feb 2004, Tim Petlock wrote:
 
[...]
 I dial 800 numbers all the time from my Nufone account 
 without problem. 
 Hell, my DID through Nufone -IS- an 800 number!
[...]

Of course you're aware that a DID and call termination are completely
different things that have nearly nothing in common.

Daryl
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RE: [Asterisk-Users] Dialing 800 numbers with VOIP

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Rob Fugina
 Sent: Tuesday, February 10, 2004 2:28 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP

 Is this really a company, or is NuFone in some guy's basement?

And being in someone's basement has exactly WHAT to do with being a real
company?

Get with the times.  You don't spend money on offices when you don't
need them.  Nor on datacenters.

I started, what I can only assume you would consider to be not a real
company, in my basement, about 8 years ago.  It was sold for 7 figures 4
years ago.  I'm in the process of doing the same, this time from my
attic.

I'm tired of the NuFone bashing.  If you don't like a company, don't use
their services.  If you want to bitch and complain about service
providers, find another list (and come up with a better argument).  This
one is about Asterisk.  Not the services you can use Asterisk with.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] Can asterisk make a call to a phone?

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 John Chambers
 Sent: Monday, February 09, 2004 12:21 PM
 To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Can asterisk make a call to a phone?
 
 
 Newbie question coming up ...
 
 Is it possible to use the asterisk to initiate a call to a phone?

Yep...it's on The Wiki at http://www.voip-info.org.

Specifically, I think
http://www.voip-info.org/wiki-Asterisk+auto-dial+out is what you're
looking for.

Daryl
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RE: [Asterisk-Users] PCI expansion slots.

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
 Sent: Saturday, January 31, 2004 10:05 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] PCI expansion slots.
 
 
 
 Hello,
 
 Did anyone use PCI expansion slots such as:
 
 http://www.cyberresearch.com/store/product/311.2.htm
 
 I want to know how well does it work with Asterisk FXO/FXS 
 cards?  Also, does FXO/FXS drivers work automatically 
 (meaning seemlessly recognize the expansion slots) without 
 any Power/Bandwidth/Interrupt issues?
 
 Any alternative or information about working (or not working) 
 baords would be highly appreciated.

That's not a PCI Expansion Slot.  It's a passive backplane, designed
to host a single-board industrial type machine.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
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RE: [Asterisk-Users] determining legal VoIP service

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Walker Haddock
 Sent: Friday, January 30, 2004 5:52 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] determining legal VoIP service
 
 
 Can anyone recommend who we can consult with that could 
 provide advice on the legality of a proposed VoIP service.  
 Specifically, we would provide VoIP termination in the USA to 
 clients in Spain, Nigeria and Guana.  The termination service 
 would connect the VoIP clients to the PSTN through a carrier 
 like MCI, Verizon, etc.  The calls placed would connect 
 anywhere in the world via the USA carrier.

If you're interested in receiving traffic for those locations, you could
talk to ITXC.  They mostly sell H.323 termination though other people's
POP sites around the world.  They may need more termination in those
areas.

If so, they are very well versed in this type of thing, and could
probably help you out (not just in getting your proosed plan running,
but possibly making a good bit of money on top of that).

I don't know anyone in the reseller/sales division, only their
engineers, but you might want to give them a try.  The could become a
very good customer/peer of yours if you happen to be terminating in the
right spots.

http://www.itxc.net

FYI, I don't see much on their web site about becoming a SNOC as they
call it, but I'd try just giving sales or a general number a call and
see who you can get transferred to.

Best of luck,
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
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RE: [Asterisk-Users] Re: Digium X100P for $43

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Sean Cheesman
 Sent: Wednesday, January 21, 2004 11:04 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Re: Digium X100P for $43
 
 
 for the record, mine has the same fcc id number as the 
 Digiums.  Is this typical for copied hardware, or is there 
 something a little fishy going on here?

No, nothing fishy.  It's a WinModem.  Digium didn't make it to begin
with.

It's commodity hardware.

You can get them for $14-19 a piece.  But that's just not the right
thing to do.  Asterisk development is paid for in part by sales of this
hardware.  Buy it from Digium, and you get support as well.

I had a problem compiling the zap drivers when I got mine.  When I
called, the phone was picked up immediately, by a real person who knew
exactly what they were talking about.  Digium support actually SSHed
into my box and fixed it/showed me what I was doing wrong.

The support is well worth the price, especially if you are building a
production server.  Or if your time is worth anything at all for that
matter.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
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RE: [Asterisk-Users] ADSI phone vs. IP phone (and proper implementation thereof)

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ray Burkholder
 Sent: Monday, January 19, 2004 7:38 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] ADSI phone vs. IP phone (and 
 proper implementation thereof)
 
 
[...]

 I'm wondering if what you say is actually true.  According to 
 recent media 
 releases, Cisco has shipped over 2 million of their IP 
 phones.  They must be 
 doing something right.
[...]

Yes, they are marketing well, and the phones work just fine.  But what
does the number of units shipped have to do with anything?  I've got a
dump truck load of 1721/VPN-K9s with ADSL cards that STILL have an open
bug after almost six months (which causes blocking on the ATM port,
rendering the router unable to pass traffic).  Does that mean they are
perfect?  No.

I'd go as far as saying that the 7960s are better than that, as they
work very well.  Until you try to use the built in switch and hit the
right conditions.

[...]
 Voip quality is not necessarily about bandwidth (because it 
 works on T1 data 
 lines as well as GB ports), but about instantaneous 
 bottlenecks in the 
 network.  These instantaneous and random bottlenecks can 
 occur in the cad 
 environment mentioned.  But with appropriate COS (layer 2) 
 and TOS (layer 3) 
 settings in the phones, switches, and routers, these 
 bottlenecks become non- issues.
[...]

That's VoIP 101.

The real issue is that the phones crash/reboot/degrade under high pps on
the switch.  Probably because of all of that processing for VLANS and
switching taking place on the same processor as the phone (just a guess,
I have no idea of the internal design).

Go get yourself a nachi-style worm, or other high-pps type app and put
it on a reasonable well-powered machine on a 7960.  Crank up the packets
and try to make phone calls.  Then we'll talk again.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 

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RE: [Asterisk-Users] ADSI phone vs. IP phone

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of PJ
 Sent: Tuesday, January 20, 2004 5:09 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone
 
 
 On Mon, 19 Jan 2004, David Gomillion wrote:
 
  Andrew wrote:
  
  First, what's wrong with PoE?  Is it any worse than 
 installing tons of 
  channel banks?
 
 Can anybody recommend a good PoE product?  I am interested in 
 getting that implemented.

You need to be more specificPoE isn't all standard.  As is par for
the Course, Cisco has their own.

So If you're talking about 79xx's, I can definitely recommend any of the
PoE blased for the Cat 4500 and 6500 series.  Just make sure you have
enough wattage coming form your power supplies (I had to go to 220v on
one after loading it up with PoE blades).

For smaller wiring closets, the Cat 3524-PWR-XL works great.

And if you also have a Cisco wireless infrastructure (AiroNet 350 and
newer) you can power those with the same hardware.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] Power Over Ethernet for *any* ethernet switch (or hub); product idea

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Ken Alker
 Sent: Tuesday, January 20, 2004 3:59 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Power Over Ethernet for *any* 
 ethernet switch (or hub); product idea
 
 
[...]
 Assume I have a non-POE switch with 24 RJ-45 (ethernet) 
 ports.  I design a 
 1U box that can be mounted just above/below the non-POE 
 switch, call it a 
 POEI (POE inserter).  This box has 48 RJ-45 ports, 24 
[...]
 Are POE switches expensive enough to warrant manufacturing 
 above? If not, is there a case for not having to swap out all 
 of ones existing 
 switches?
[...]
Depends on what expensive means, and whether your switces are due for
replacemtn or not.  And what you intended to replace them with not
counting PoE.  The difference between Catalyst 2950-XL-24s and
3524-PRW-XL's is about $300.

The difference on a large Catalyst switch is about $5-10/port if I
recall correctly from my last deployment.

 Does something like this already exist for cheap?
Yes.  Several.

 If so, is it any good?
Yes.  Many work just fine.

 If so, does it need more features?
To do what?  It's called a mid-span power injector.  The ones I've seen
do that and nothing else.  I'd say they are living up to their task.

 If not, would you buy something like this?
 If so, what features have I missed?
 If so, what is it worth?
Google the rest of your answers.  You're about 6 years too late to catch
the first run of this train.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] ADSI phone vs. IP phone

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steven Critchfield
 Sent: Tuesday, January 20, 2004 1:09 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] ADSI phone vs. IP phone
 
[...]
  You need to be more specificPoE isn't all standard.  As 
  ^^^
[...]
 PoE has a standard. But some manufacturers either put their 
 product out before the standard was fully agreed upon, or ignore it.
[...]

Yes, note the highlighted section of what I said.

When a market-shareholder as large as Cisco has their own
implementation, it makes the standard not so standard anymore.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] DTMF A-D

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Ken Alker
 Sent: Tuesday, January 20, 2004 1:37 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] DTMF A-D
 
[...]
 I've know about DTMF A-D for 20+ years now, but have never 
 heard anyone 
 mention it before, or use it, for that matter (except in old silver 
 boxing in the bad ol' days).  Can you elaborate upon how you'd take 
 advantage of DTMF A-D, how you'd produce the tones (are these 
 standard 
 now?), and what exactly you mean by muting from the far-end?
[...]

Let me take a crack at this one:
4th column tones have been standard for a very long time, and are often
used for conference control functions.  I have several phones that have
A-D on them.

Which I'm sure is where the original poster would want far-end muting
capability (no need to blast DTMF into a conference call for all to hear
when you are performing control functions on the call).

Whether this is the best way to do things today, especially in hybrid
environments is another discussion.  My personal preference is to see
those nasty things (all in-band signaling, for that matter) die off in
favor of more modern control methods made much more accessible with SIP
phones and large, programmable displays with soft buttons.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] ADSI phone vs. IP phone

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dustin Goodwin
 Sent: Monday, January 19, 2004 11:18 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] ADSI phone vs. IP phone
 
 
 Why wouldn't you just use your existing Ethernet 
 infrastructure putting 
 the  IP phones inline between the wall jack and the PC? There are a 
 number of IP phones that have builtin switch/hub that allows 
 the PC to 
 daisy chain off the IP phone.

Probably because it's well known that these setups are prone to failure
of either the PC's connection, the phone's connection, or degredation of
one/both.  It also breaks switch envirenments where spanning-tree
portfast is enabled (not as big of a deal if the deployment is in
concert with the infrastructure group, as it should be).

Vendors should NEVER have implemented this functionality into phones
unless it was working under all conditions.  Personal experience shows
that it is most definitely not on Cisco and 3Com products.  Others have
told me their stories with other manufacturer's equipment.  None of it
was good.

It's not a production-stable way to deploy phones.  Period.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
Fax: +1 508 526 8500
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RE: [Asterisk-Users] New sounds also now in CVS

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Sunday, January 18, 2004 11:22 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] New sounds also now in CVS
 
[...]
 The index for each topic could be a text file with a list of 
 phrases with 
 their corresponding file name. So there would be as many 
 files (indexes) as 
 catogories (ie Weather, monitoring, etc). When an audio clip 
 was added it woud 
 be added to one or more of these index files.  
[...]

And/or, all sound files in one directory, with a separate directory for
each topic consisting of symbolic links to the real sound files.

That's how I currently handle things on my systems.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
Tel (UK): +44 208 792 6813 x235
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[Asterisk-Users] Disturbing trend of * production boxes that shouldn't be

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Gary Franczyk
 Sent: Thursday, January 15, 2004 10:37 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Basic Asterisk capabilities question
 
 
 Whaaa?? So, to allow 24+ lines of dial in access, how would I 
 configure it?
 
 Would I need to purchase or lease a voice-over-ip box to 
 connect our T1 or phone lines into?  And then from there send 
 the VOIP to the linux/Asterisk box for recording?
 
 (forgive me, Im new to telephony, but I need to make this work) :-)

This is a disturbing trendpeople who don't know much about Asterisk
and/or Linux and/or telephony who Need to make these things work or
need to know how to update [their] production box installation...sorry
I don't know Linux at all.

Asterisk is great, but to maintain it and especially to repair things
when they've gone wrong, you need to know what you're doing.  Its your
job (the people I'm talking about know who they are), so do as you wish,
but I sure would install something I know little or nothing about and
call it production.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
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RE: [Asterisk-Users] ultra-cheap asterisk box

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Roy Sigurd Karlsbakk
 Sent: Thursday, January 15, 2004 11:08 AM
 To: Asterisk Users
 Subject: [Asterisk-Users] ultra-cheap asterisk box
 
 
 hi all
 
 what about this...
 I just put together a box on a web shop (komplett.no) that 
 will cost me NOK ~1850 ( 216) plus a small 50 drive and 
 cables, so say 300. This consists of a cheap MB with a duron 
 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will 
 finish off the zaptel-driver soon). This is all in a cheap PC case.
 
 What do you think? Should this be doable? as a product? With 
 only IP phones and potentially a fax solution? any ideas?

I've got one system with 10 IP phones + SIP term + 2 FXS + 4 FXO running on a P700 
with 256 MB RAM.  It works just fine, and the CPU is rarely over 40%.

Sounds like that box will work from a capability standpoint.
Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
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RE: [Asterisk-Users] re hardware requirement - asterisk

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Chris Albertson
 Sent: Thursday, January 15, 2004 12:40 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] re hardware requirement - asterisk
 
 
 
 I don't think 10BaseT can run full duplex.  I could be wrong 
 but I don't think so.

Where'd you get that idea from?  A 10-Base-T connection to a switch port
most definitely will (and should) fun full duplex.

 But why does it matter?  A single VOIP connection will not 
 even use 1% of a simplex 10BaseT.  Simplex 100BaseT should be 
 able to handle dozens and dozens of calls

Properly configured, yes.  I don't know the details of your issue, but
I've seen more shoddily auto-detected connections that I care to
remember (3Com cards on Auto - Cisco Catalyst on Auto anyone?).  Lock
the speed/duplex on the switch and the server, and check for collisions,
etc. on the port.

Daryl G. Jurbala
BMPC Network Operations
Tel (NY): +1 917 477 0468 x235
Tel (MI): +1 616 608 0004 x235
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RE: [Asterisk-Users] More words for Allison

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 [EMAIL PROTECTED]
 Sent: Sunday, January 11, 2004 8:39 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] More words for Allison
 
[...]
 snip
  knots per hour
 
 I'm a land-lubber, but I think knots is a speed unit (like 
 Miles Per Hour), so I think you want knots here, not knots 
 per hour, if you are talking wind speed.
 
[...]

Then stick to being a land lubber.  Because you're wrong.

A knot is a unit of linear measurement.

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

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RE: [Asterisk-Users] More words for Allison

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Dave Cotton
 Sent: Monday, January 12, 2004 11:24 AM
 To: Asterisk List
 Subject: RE: [Asterisk-Users] More words for Allison
 
[...]
 
  A knot is a unit of linear measurement.
 
 Perhaps you're both wrong or right :)
 
http://www.yourdictionary.com/ahd/k/k0092800.html

[...]

Truein common usage a knot means either nautical mile or nautical
miles per hour depending on the context.

That was a pre-coffee post.  I can take only partial responsibility. ;)

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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RE: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of David Burr
 Sent: Sunday, January 11, 2004 4:31 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] CONTEST: Top Posters win 80G Hard Drive
 
 
 We have a new contest starting today!
 
 The first three members to post 300 messages at 
 http://www.asterisk.bz 
 will win a _80Gig Hard Drive!_
 
 Its 
 quite simple. Messages must be asterisk related.

I can guarantee you that very few people who know anything about * will
be posting to that site.

It's a horrible interface, as nearly all forums that try to duplicate
mailing lists are.  Do you really think Linux/UNIX CLI guys want to deal
with a web site where insert their own favorite mail reader here will
do just fine?

Highly technical forums, especially those related to primarily CLI
tools, fail miserably.

Have you not been reading this list as to the feelings of the regular
contributors and answer-providers on this issue?

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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RE: [Asterisk-Users] Mailing list growth

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Philipp von Klitzing
 Sent: Saturday, January 10, 2004 10:35 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Mailing list growth
 
 
 - asterisk-users: VoIP and Asterisk in general (including newbies)
 - asterisk-tdm: Use if part of your problem/question involves T1/TDM
 - asterisk-biz: new topics, not yet really covered on -users
 
 Effects:
 - newbies only need to subscribe and read a lower volume -users
 - all readers have the same amount of traffic, but get some nice 
 filtering help at least

Reasonable, but may need some serious topic policing at first (requiring
multiple list admins per list), again due to the fact that people often
will not know where their problem lies.

Also, just as an example.the VoIP list would have discussions on it
like the recent calling card appwell, that doesn't sounds newbieish
at all.

Has anyone actually taken the time to do a message/category
classification and breakdown to see if the proposed split even makes
sense?  Would we end up with 10 messages a day in -biz, 25 or so in -tdm
and 100 in -users?

 As Robert pointed out LISTSERV has some nice topic features 
 that could 
 help, however the license ist costly (we have two LISTSERVs 
 running). Let 
 me add, though, that besides topic management LISTSERV can 
 also provide 
 super lists that are great to fight cross-postings - super 
 lists group 
 one or more normal lists or super lists. My guess is that 
 there are other 
 MLMs out there that have similar features.

LISTSERV is evil, and yes, there are (listserv is evil mostly because of
the abhorrent cost of something that is available via open source/free
alternatives and a couple of perl/awk/sed scripts).

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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RE: [Asterisk-Users] Mailing list growth

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Philipp von Klitzing
 Sent: Friday, January 09, 2004 6:52 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Mailing list growth
 
[...]
 higher-level implementation list that deals specifically with 
 channelbanks  T1 issues (=larger installations). VoIP will remain on 
 asterisk-users.
[...]

That doesn't quite sound right.  Maybe it is from your perspective, but
are you telling me that the NOCs with 80+ 7960's running VoIP don't
count as a large installation?

Of course, the term large is also relative.  A 4-port T1 card on its
owneven 2 or 3 of them, could never by any stretch of my imagination
be considered a large installation..but I deal with (among other
things) Definity's that service near entire buildings in mid-town
Manhattan with multiple DS3sso it's all relative.

The problem with splitting VoIP and T1/TDM/whatever you want to call it
is that the crossover is huge, and where the problems lie often aren't
clear to those looking for help.
Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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RE: [Asterisk-Users] Screen Pop Remote Agents = Telemarketing

[daryl]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of empire
underground
Sent: Friday, January 09, 2004 1:32 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Screen Pop  Remote Agents
 can I put a .csv file in the sql DB and have it dial from there? and
will I be able to set a
 Dial Plan to only call certin area codes? stuff like that. The reason
I ask all this is because
 all of these over priced dialers do just that. Also can Asterisk be
set with the FTC laws to 3%
 droped call ratio?
 If all of the questions I have asked here have allready be answered
some point in time... Can
 someone pl ease point me in the right direction to get all the
answers.

So you're setting up a telemarketing rig?  That's certainly not the kind
of thing I'd expect to get much help or sympathy for ANYWHERE other than
in telemarketing circles.

I think the cost of a proper commercial predictive dialer would be
relatively cheap after already having sold your soul.

Daryl
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RE: [Asterisk-Users] Cisco to Cisco - poor quality

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Terence Parker
 Sent: Sunday, January 04, 2004 8:29 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality
 
 
 Thanks for the replies.
 
 My cisco firmware is only POS3-04-2-00, though it is SIP. It 
 used to work fine under vocal though - which was strange. Is 
 this definitely nothing to do with asterisk? I do note 
 however that my firmware is fairly old... except cisco aren't 
 exactly generous with firmware upgrades.
 
 I have tried both g729a (default on my phone) and g711ulaw 
 with no success. But i'll have another fiddle and try to get 
 it to work.

How are the phones talking to each other?  Directly, or through
asterisk?  (canreinvite=what? in the sip.conf for each of them?).

What I'm trying to get at here is, it is a problem between the phones,
or are you having a problem possibly with the asterisk box?  Some other
things to know: are you running voicemail yet?  If so and you can dial
into it from either of the phones, how does it sound?  If not, how about
anything from the * boxlike the demo annoucment stuff?

Daryl
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RE: [Asterisk-Users] Sun Servers with UltraSparc Processors

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Tilghman Lesher
 Sent: Sunday, January 04, 2004 9:58 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Sun Servers with UltraSparc Processors
 
 
[...]
 
 You shouldn't face any problems with endianness.  In fact, 
 the core code should probably work right out of the box.  

Not as of 3 months ago, the last time I tried.  Test platform was Debian
running on a Netra T1 120.

I did little to try to go further, as I wasn't sure how I was going to
get a timing source even if * did compile.

 And I think it's excellent that you're thinking about doing 
 this in the lab, first, and not buying your production 
 machines without testing, first.  There've been a few users 
 who jump a little too fast, then get disappointed when their 
 first machines didn't always perform to their expectations.

I second that.

And I'd love to get * working on a Sparc.  As a matter of fact, I've for
a SunFire V120 doing absolutely nothing.  (along with a few T1's and
soon to be an E220r...I'm phasing sun out of my NOCs due to insane
new hardware costs, wonkiness and expense of Solaris-based management
platforms (can you say dependency hell to the 1000th degree?) and how
ridiculously easy VMWare makes managing multiple low-usage
installations).

Maybe I'll give it a shot.  But I'm not a developer by any means.  Maybe
if one or more of the * devs and/or hackers want to help on this, I'll
consider providing test boxes hosted at on of my NOCs.  Chime in if
you're interested ([EMAIL PROTECTED]).

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp 
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RE: [Asterisk-Users] Re: Sip phones on the same extension?

[daryl]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Buhrow
Sent: Thursday, December 25, 2003 2:15 PM
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Sip phones on the same extension?


Hello.  I think I understand your suggestion, but don't
understand how that's any different than the one I came up with.  What I
want, is to be able to define a specific extension, and then have any
external SIP phones 

[...]

The difference is, his suggestion works.  Yours doesn't.

If you register multiple SIP devices in the way you suggest, only one of
them will ring.  It appears to me that the one that is fastest to
respond will work, but I only tried the setup briefly before doing a bit
of research that told me it wasn't the way this is done in *.

Daryl
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RE: [Asterisk-Users] Audio format for announcements

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Sean Adams
 Sent: Monday, December 22, 2003 10:50 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Audio format for announcements
 
 
[...]
 2) For my internal SIP phones, I don't care about bandwidth 
 usage. What 
 settings will give the best sound quality?  Does the protocol (or for 
 that matter, any particular brand of phones) support uncompressed or 
 very high bit rate audio for intra-pbx calls?

Use g.711ULAW.  I belive it is about an 87k uncompressed stream.  Sounds
better than toll quality to me.
Daryl
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RE: [Asterisk-Users] Headless Linux system for Asterisk

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Michael Welter
 Sent: Thursday, December 18, 2003 5:03 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Headless Linux system for Asterisk
 
 
 Because of space limitations and because of the location of the 
 punch-down blocks, my * server is located on the shelf in a 
 coat closet. 
   Sadly, there is not enough space (or ventilation) for the 
 monitor and 
 keyboard.  This will all change when we move to new quarters, but...
 
 Does anyone have experience running Linux/Asterisk without a monitor? 
 What, if any, are the issues?

I would doubt that many real installations have monitors attached.

And whether it works or not has nothing to do with the OS or any
applications running on the machine.  It is strictly a hardware support
issue.  Most equipment should have no problems without a mouse or
keyboard if properly configured.  Most hardware can't even detect if a
monitor is attached or not.

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp
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RE: [Asterisk-Users] Re: * with RADIUS

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jeremy McNamara
 Sent: Thursday, December 11, 2003 2:19 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Re: * with RADIUS
 
[...]
 
 Explain why you think you really need RADIUS Accounting?   
 Why not talk 
 right to the database itself and save yourself that unneeded 
 complication and points of failure.
 

I know this has come up before, and in a perfect world, where * was the
primary app, you don't need RADIUS.  In enterprise environments where
RADIUS accounting is already embedded into other aspects of the
workflow, it would be beneficial.

Understand* boxes are in real live actual production now.  Once you
leave the vacuum of the lab, there are going to be things like this that
come up.  And many will be for good reasons.  Others will be for crappy,
legacy reasons.  Both scenarios are valid in the real world.

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp
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RE: [Asterisk-Users] External Email Notification -2

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of Kevin
 Sent: Wednesday, December 10, 2003 5:06 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] External Email Notification -2
 
 
 I was looking for an assistance to help with the 
 configuration of the email on the unix server not an expert 
 on forum postings. 

Then you came to the wrong place.  If you need help configuring email on
a unix server, try the support forum for the mailer installed on your
machine.  It's not an on-topic discussion here.

After you have that working, I'm sure many people on this forum will be
more than happy to assist you in getting * to use a properly functioning
mailer to send voicemail notification.

Here's more unsolicited forum posting advice: the attitude you are
taking in the above quoted post does not inspire people to help you in
any way.

Daryl G. Jurbala
BMPC Network Operations
Tel: +1 215 825 8401 x235
Fax: +1 508 526 8500
INOC-DBA: 26412*DGJ

PGP Key: http://www.introspect.net/pgp
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[Asterisk-Users] Init.d script was: Operating environment for *

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steven Critchfield
 Sent: Thursday, December 04, 2003 4:57 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Operating environment for *
 
  3) How do you handle crashes (murphy -will- visit you some day)?
 
 There is a script that I picked up from Tilghman that runs 
 out of /etc/init.d that will continually restart asterisk 
 unless it exited with a return of 0(one of the stop 
 commands). It also sets it up so it dumps core in a 
 configured directory and can email you that it has done so.  
[...]
Steven Critchfield  [EMAIL PROTECTED]

Steven...where can I find this script, or can you forward a copy to me?

Sounds like exactly what I need at this point.

Thanks,
Daryl
[EMAIL PROTECTED]
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RE: [Asterisk-Users] PREPAID APPLECATION

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of PJ Welsh
 Sent: Tuesday, December 02, 2003 9:39 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] PREPAID APPLECATION
 
 
 It is a shame that within a couple of hours they can tell you 
 to remove helpfull documentation, but not (seemingly) help 
 answer questions regarding there Cisco stuff on this list. I 
 think Cisco must have their priorities mixed up!
 
 Just my opinion... which also means I won't support a company 
 like that... so I won't buy their products... 

No, Cisco has their priorities just fine.  Companies are in business to
make money.  Not to give things away.

When I have a problem with a piece of Cisco equipment, it is answered
promptly and accurately, nearly 100% of the time.  I have SmartNet on
all of the devices for which I expect this service.

Cisco documents are property of Cisco.  Many of them require a CCO
account to access, and there are varying levels of CCO access.  Many
newer technologies are initially available to all with a CCO login
until their maturity and complexity reaches a point where Cisco makes a
specialty for them, at which time those documents and new ones on the
subject are sometimes no longer available to just anyone with a CCO
login.  Cisco also maintains and updates their documents on an as-needed
basis.

Storing copes of their documents on your own web site for public use
defeats their ability to do all of these things.

And if you want to argue that much of this is done just to charge you
more money, you are correct.  Cisco is an enterprise infrastructure
company.  Not a home user/home office/small office outfit where you can
call up and talk to a 17 year old with a script for help when you have
a problem.  To get real support costs money.  Example: Digium.

Daryl
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RE: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD

[daryl]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Roy Sigurd Karlsbakk
 Sent: Friday, November 28, 2003 6:50 AM
 To: Asterisk Users
 Subject: Re: [Asterisk-Users] Re: ASTERISK WITHOUT ANY CARD
 
 
  CAN I USE/COMPILE ASTERISK
  without any telephone/sound card?
  I only want to use it as a IP PBX.
  
  YES
  you can.
 
 how about IAX2 trunking? does this work with ztdummy?

I was using both IAX2 trunking and MOH before getting my zap devices,
and I never had any luck with ztdummy.
Daryl
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