Re: [Asterisk-Users] UPS for Asterisk

[Peter Svensson]

On Sun, 23 Jan 2005, Andrew Kohlsmith wrote:

> Why would the heads come in contact with the platters on a powerfail?  The 
> arms are very rigid -- the heads only float a few thousandths of an inch over 
> the platters -- something that I don't believe has anything to do with the 
> platters spinning (that may *help* but I don't think the heads will contact 
> the platters if they're not spinning) and besides -- any drive manufactured 
> in the last 5 years will autopark on power fail...  There's an awful lot of 
> energy stored up in the spindle motor that is used to slam the heads into the 
> parking zone...

Actually, the only thing that keeps the heads off the platter is the fact 
that they are spinning. The movement of the platters cause an airstream 
which the heads float on. This airstream is what keeps the heads at just 
the right distance. The arms are not very rigid at all in the axis 
direction of the disks. This has been the standard design in hard disks 
since a very long time.

The comment about autopark is correct. Actually, with the voice coils used 
on modern disks the energy needed to retract the heads is already stored 
in the return spring. The platter energy is sometimes used to complete 
any sector write that is in progress. Some hard disks did not do this and 
those generated bad sectors every time they were powered down in 
mid-write.

Peter

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: IAX Inbound Sound Quality

[Lee]

On Sun, 23 Jan 2005 01:51:56 -0500, Andrew Kohlsmith > I have *no*
issues on inbound quality with sixTel.  They *had* a problem where
> the first second of audio was cut off upon connect (Wait() did not help) but
> that seems to have been fixed.

I see this problem intermittently, typically during a long call, after
45 min. or so.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Looking for a prepaid calling card platform

[James H. Thompson]

I'm looking for a prepaid calling card platform 
that:
 
* easily scales to multiple servers with a common database 
for: redundancy, capacity, and performance
Looking to start with capacity to handle 100 simultaneous 
calls and be able to easily scale to 1000+ simultaneous 
calls.
 
* in addition to the normal anti-fraud measures, supports an 
API for easily adding new anti-fraud tests along the lines of the 
following:
 
For each new call being attempted the 
system would invoke an external authentication program and 
pass: reseller ID, card ID, time left on card, called # and calling #; 
and the history/status for the last several calls including for each call 
the called #, calling #, call duration, call timestamp and call status 
(in-progress, completed, etc).  Progrm would return: call OK, deny call 
with recording #x, invalidate card with recording #y.
 
* ability to limit calls to a maxium duration and/or to 
require periodic IVR user response to continue a long call.
 
* contolled, managed and provisioned with a web 
interface
 
* support multiple resellers, each with password protected web 
access for managing their customers.
 
* ability for customers to call an 800# to hear a 
recording giving them the user a local non-800 number they need to 
call to use the card.
 
* credit card recharge support
 
While willing to do minor customizations, would like to find 
something that is mostly install and go.
Open source would be nice, but willing to pay for a well done 
package.
 
Suggestions welcome.
 
Thanks.
 
Jim
 
James H. Thompson[EMAIL PROTECTED]
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sip - h323 translation stability & capacity limit

[Lubomir Christov]

HI John,
It also depends which H323 channel you will use for this translation.
I can recommend you to use the chan_oh323 from inAccess Networks - 
according to our experience it's much stable and bug free channel.
Our Asterisk based translation system is running much stable with 
chan_oh323 ..
The sip-h323 &  h323-sip translation now is working pretty well :)

Best regards,
Lubo
-
AppRadius Project: Full RADIUS AAA support for Asterisk PBX
http://appradius.minitelecom.org/
-

[EMAIL PROTECTED] com wrote:
Hi! All
  I would appreciate if someone could advice me on how stable is sip-h323 &  
h323-sip translation as well as how many calls can it handle when doing such 
translation.( assuming single 2.8Ghz intel processor & 1GB RAM)
Regards,
John
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Music On-Hold problem

[Doug Reid - Stormcorp]

What handset? Some such as the Planet dont work. 

>  -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] 
> Sent: Monday, January 24, 2005 1:10 AM
> To:   asterisk-users@lists.digium.com
> Subject:  [Asterisk-Users] Music On-Hold problem
> 
> My problem is: No matter what machine I install and configure Asterisk on
> it  I just can't get the music on-hold to work. Is anyone of you out there
> have such problem? If so what have you done to fix the problem? I've tried
> so far three other computers and none of them I was able to get music
> on-hold to work.
> 
> CAN PLEASE ANYONE HELP. << File: ATT00718.txt >> 
<>___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] finding current codec?

[Adam Goryachev]

See www.websitemanagers.com.au/asterisk/
It will allow anyone to contribute their tools/etc... Of course, some
things are more suitable to the wiki (eg, dialplan snippets/info/etc).

Regards,
Adam

On Sat, 2005-01-22 at 13:12 +0200, Mike Tkachuk wrote:
> Hello,
> 
> I dunno if it's really needed, we should ask Mark. Anyway I created site
> http://b2bua.berlios.de
> where I will post all my asterisk patches and applications.
> 
> On Fri, 21 Jan 2005 11:57:52 -, Muhammad Nasim
> <[EMAIL PROTECTED]> wrote:
> > Hi Mike
> >  
> > This is a damn useful app. Do you know if its been put in cvs yet?
> >  
> > Kind Regards
> >  
> > Mo
> >  
> > Muhammad Nasim
> > Telappliant Ltd 
> > Tel: 020 7043 3492
> > Int: +44 20 7043 3492
> > Main: 0845 004 4040
> > Fax: 0845 004 4041 
> > www.telappliant.com 
> > 
> >  
> >
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Andrew Yager]

On 24/01/2005, at 3:26 PM, Andrew Kohlsmith wrote:
On January 23, 2005 10:30 pm, Nick Bachmann wrote:
 HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD
manufacturers have done "head unloading" and such recently, but the 
risk
is still higher if power is suddenly lost during a write.
Why would the heads come in contact with the platters on a powerfail?  
The
arms are very rigid -- the heads only float a few thousandths of an 
inch over
the platters -- something that I don't believe has anything to do with 
the
platters spinning (that may *help* but I don't think the heads will 
contact
the platters if they're not spinning) and besides -- any drive 
manufactured
in the last 5 years will autopark on power fail...  There's an awful 
lot of
energy stored up in the spindle motor that is used to slam the heads 
into the
parking zone...
Yet it is still a problem, and still happens. In fact, I've had three 
machines in the last three months. It was enough to convince the 
cusotmers that a UPS was indeed what they needed to protect their 
investment.

A UPS is a good investment. It protects hardware against anything going 
wrong, and allows for those rare, but painful blackouts that sometimes 
occur.

Andrew
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Peculiar one way convesation fault with Asterisk.

[Voicomm User]

Dear Fellow Ast-Masters

Has any one here experienced the following issue (or similar).

Setup Description:
  - Asterix Server  - 192.168.1.10
  - Analog Phone connected through FXS to Asterisk Box - (ext) 2000 
  - SoundPoint IP 300 - 192.168.1.2 - (ext) 4004

Problem: 
I have a very basic sip.conf, extension.conf and zapata.conf. 
Now when I try using 4004 my call doesn't go through. But if 
I dial 4004 from 2000, it works like a charm. And, soon after 
I dial 4004 from 2000, if I try using 4004, it works fine too. 

Now if I leave it idle for a while and then try using 4004 to 
dial any number it doesn't work. The LCD on my soundpoint IP 
shows status as 'connecting' and then after a while gives up.
Whats even more peculiar is, if I issue 'reload' command in 
my asterisk server console it starts working fine. 

Problem solving attempts:
To find out more I started sniffing packets in asterisk server.
In ethereal this is exactly what I see,

Source Dest Info
192.168.1.2  192.168.1.10 SIP Request (blah blah)
192.168.1.10192.168.1.2   ICMP: Destination Unreachable, 
   type-3, code-10 (host 
   administratively prohibited)
192.168.1.2 192.168.1.10  SIP Request (blah blah)
192.168.1.10192.168.1.2   ICMP: Destination Unreachable, 
   type-3, code-10 (host 
   administratively prohibited)



After some reading I found out that this type of ICMP messages 
are normally sent out by the router when a host is blocked. The
document I read also adds, its mostly used in US government 
implementations(?). I haven't seen this type of ICMP message 
previously and not completely sure if I am interpreting 
it correctly. Are these messages even getting up to Asterisk 
server? If not, why not? if yes, then why is asterisk not 
accepting these messages?

Any leads or help greatly appreciated.

-r
PS: I have googled around, but to no avail.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] sip - h323 translation stability & capacity limit

[[EMAIL PROTECTED] com]

Hi! All
  I would appreciate if someone could advice me on how stable is sip-h323 &  
h323-sip translation as well as how many calls can it handle when doing such 
translation.( assuming single 2.8Ghz intel processor & 1GB RAM)
Regards,
John
-- 
___
Sign-up for Ads Free at Mail.com
http://promo.mail.com/adsfreejump.htm

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] zaprtc from bristuff? not there?

[Spencer Nassar]

I'm trying to take advice to use "zaprtc from bristuff" (from both a 
posting here and references on voip-info) because I have a 2.4 kernel 
SMP machine.

I've downloaded and installed bristuff-0.2.0-RC3a and now have the 
modules zaphfc and zaptel loaded.

Running meetme says the extension is invalid (I've double checked 
meetme.conf and extensions.conf).

There is no zaprtc version in that bristuff package (I thought there 
would be based on other posts).  Trying to load zaprtc 0.0.1 gives the 
same error as the standard asterisk distribution.

Do I download something more recent than zaprtc 0.0.1?  Can anyone tell 
me were (no luck googling)?  Other insights?

Thanks much!
Spencer
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Andrew Kohlsmith]

On January 23, 2005 10:30 pm, Nick Bachmann wrote:
> >  HDDs don't fail because they lose power.
> Unless the heads crash, which can happen if power fails. I know HDD
> manufacturers have done "head unloading" and such recently, but the risk
> is still higher if power is suddenly lost during a write.

Why would the heads come in contact with the platters on a powerfail?  The 
arms are very rigid -- the heads only float a few thousandths of an inch over 
the platters -- something that I don't believe has anything to do with the 
platters spinning (that may *help* but I don't think the heads will contact 
the platters if they're not spinning) and besides -- any drive manufactured 
in the last 5 years will autopark on power fail...  There's an awful lot of 
energy stored up in the spindle motor that is used to slam the heads into the 
parking zone...

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Music On-Hold problem

[Manjit Riat]

Title: RE: [Asterisk-Users] Music On-Hold problem






It should work right off the install..

Make sure you have MPG123 installed and running.

_
From: Computer Onsite Support [mailto:[EMAIL PROTECTED]]
Sent: Sunday, January 23, 2005 3:10 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Music On-Hold problem

My problem is: No matter what machine I install and configure Asterisk on it  I just can't get the music on-hold to work. Is anyone of you out there have such problem? If so what have you done to fix the problem? I've tried so far three other computers and none of them I was able to get music on-hold to work.

CAN PLEASE ANYONE HELP.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Michael 'Moose' Dinn]

> And, in fact, some drives *do* have problems with sudden outages.  Some 

Relative to the cost of a cheap UPS, downtime is much much much more
expensive. You can power pretty much any single server you want for ~$150 CDN,
and shut it down cleanly when the power goes out. Compare $150 with the cost
of rebuilding the machine and it's money well spent. That doesn't even
consider the screaming customers.

Every machine I have in the field with a hard disk has a UPS - sometimes only a 
350VA UPS,
but a UPS none the less.  The machines that boot from CF cards are a different
story...

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Scott Laird]

On Jan 23, 2005, at 7:30 PM, Nick Bachmann wrote:
> As I understand, if HD activity is minimal, the probability of HD
> failure is significantly reduced.
 HDDs don't fail because they lose power.
Unless the heads crash, which can happen if power fails. I know HDD 
manufacturers have done "head unloading" and such recently, but the 
risk is still higher if power is suddenly lost during a write.
And, in fact, some drives *do* have problems with sudden outages.  Some 
recent IBM drives will interpret sectors that were only partially 
written when the power failed as bad blocks and refuse to read or write 
to them when the power comes back on.  I wouldn't be surprised if other 
drives have similar problems.  FWIW, we have a drive in a test system 
at work that started developing problems immediately after a power 
outage a couple months ago.  It might be just a coincidence, but the 
timing was right--the power went out in the afternoon, and the evening 
SMART media check found bad sectors.

Scott
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] here's my IAX callthrough app and some questions about problems I have.

[Brian Dingman]

Did you ever get DTMF to work reliably with LiveVoip. I am having the
exact same problems.


On Mon, 17 Jan 2005 20:22:30 -0500, Jess Coburn <[EMAIL PROTECTED]> wrote:
> Hello all,
> 
> What my app does is accepts a call in on a Dial-In Number (DID) via
> IAX, and then prompts the caller for the top secret password (123) and
> then authenticates the user and prompts them to dial in the number
> they'd like to call. Once they press pound after dialing in the number
> it will read it back to them, if they press pound it will attempt to
> connect via the second IAX provider, if they press star it will allow
> them enter in the number over again.
> 
> Now here's the problems and questions:
> 
> 1. DTMF detection seems flawed, sometimes it's dead on but alot of
> times it will see a single keypress as multiple keypresses. So I may
> press 561 but it will see 51 and all three keypresses are about
> the same length. Is this unique to my case or do you others see this
> too. I suspect it's due to either background noise or maybe
> packetloss? Any ideas on how to clean this up?
> 
> 2. The only way I can get the app to fire off is if I put the
> extension mapping in as _NXXNXX,1,CMD I'd like to use s,1,CMD but
> I don't know what I'm missing here or doing wrong.
> 
> Below are a copies of my extensions.conf file and my iax.conf file.
> 
> Regards,
> Jess
> 
> extensions.conf
> file-
> 
> [general]
> static=yes
> writeprotect=no
> 
> [globals]
> ${OUTGOING-NUM}=
> 
> [arbitrary-in]  ; <-- Should match the context you have
>; under [incoming] in iax.conf
> exten => _NXXNXX,1,Answer
> exten => _NXXNXX,2,Background(vm-password)
> exten => _NXXNXX,3,Authenticate(123)
> exten => _NXXNXX,4,Playback(beep)
> exten => _NXXNXX,5,SetVar(NR=)
> exten => _NXXNXX,6,Goto(testdtmf|s|1)
> 
> ;
> ; This context is used by the sample [arbitrary-name]
> ; context above to read back each digit you press.
> ;
> [testdtmf]
> exten => s,1,SetVar(NR=)
> exten => s,2,Background(pls-entr-num-uwish2-call)
> exten => s,3,Background(and-prs-pound-whn-finished)
> exten => s,4,Background(beep)
> exten => s,5,WaitExten(10)
> exten => _x,1,SetVar(NR=${NR}${EXTEN})
> exten => _x,2,NoOp(${NR})
> exten => _x,3,Goto(testdtmf|s|5)
> exten => _#,1,Goto(verifynumber|s|1)
> exten => i,1,Goto(testdtmf|s|1)
> exten => t,1,Hangup
> 
> [verifynumber]
> exten => s,1,Background(you-dialed)
> exten => s,2,SayDigits(${NR})
> exten => s,3,Background(if-correct-press)
> exten => s,4,Background(pound)
> exten => s,5,Background(otherwise-press)
> exten => s,6,Background(star)
> exten => _#,1,Background(pls-wait-connect-call)
> exten => _#,2,Dial(IAX2/[EMAIL PROTECTED]/${NR},30)
> exten => _#,3,Background(something-terribly-wrong);
> exten => _#,4,Background(goodbye)
> exten => _#,5,Hangup
> exten => _*,1,Goto(testdtmf|s|1)
> 
> iax.conf file 
> --
> ; iax.conf
> 
> [general]
> 
> ${INCOMING-USR}=SECRET-USERNAME
> ${INCOMING-PWD}=SECRET-PWD
> ${LIVEVOIP-SVR}=217.160.244.186
> 
> bandwidth=high
> disallow=lpc10
> jitterbuffer=yes
> dropcount=2
> maxjitterbuffer=500
> maxexcessbuffer=80
> minexcessbuffer=10
> jittershrinkrate=1
> 
> register => ${INCOMING-USR}:[EMAIL PROTECTED]
> tos=lowdelay
> 
> [incoming]
> ; this is the incoming IAX provider
> type=user
> secret=ITS-SECRET
> deny=0.0.0.0/0.0.0.0
> permit=217.160.244.186/255.255.255.0
> context=arbitrary-in
> 
> [outgoing]
> ;this is the outgoing IAX provider
> type=peer
> host= 216.118.117.46
> secret= ITS-SECRET
> auth=md5
> notransfer=yes
> context=default
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Festival

[Gary]

On Mon, 24 Jan 2005 14:57:06 +1300, Matt Riddell wrote:

>Howard Lowndes wrote:
>> Is it possible to get the Festival command to read the text from a
>> system file rather than having it input as a text string?
>> 
>> Is this a case of having to use AGI, or is there a simpler way?
>
>Most people would use AGI for that (combined with the text2wave or 
>whatever program).  In fact there may even be an example on the wiki.

I might also add that if you look in the wiki for cepstral as well some
good examples.

And cepstral voices sound much nicer than festival :-)
.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk 1.0.5

[Russell Bryant]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everyone,
As you know, we released Asterisk 1.0.4 earlier this week.  However,
there was a callerid bug in chan_zap that has caused us to go ahead and
make another release.  Asterisk 1.0.5 is available at all of the usual
locations.
I'm sorry for any inconvenience this may cause.
Russell Bryant
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.0 (GNU/Linux)
iD8DBQFB9GyrrwroOS5t/FoRAqwTAJ96XvSW7QzctTkV+MBh+nLkfe8RgQCeO8Ep
68u3BuZgT9jgANDceGT1u1k=
=aghw
-END PGP SIGNATURE-
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Nick Bachmann]

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

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

Nick
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Music On-Hold problem

[tom]

i have had some problems with music on hold. some of the handsets havnt been 
able to put people on hold...

When using a grandstream 101 i push hold it puts the other end on hold but 
doesnt play the music. although when i do it with a x-lite client it does put 
the other end on hold and stats the music.

i havnt been able to get the grandsteam clients to play the hold music yet 
though :/
 Computer Onsite Support <[EMAIL PROTECTED]> wrote :

> My problem is: No matter what machine I install and configure Asterisk on it
> I just can't get the music on-hold to work. Is anyone of you out there have
> such problem? If so what have you done to fix the problem? I've tried so far
> three other computers and none of them I was able to get music on-hold to
> work.
> 
> CAN PLEASE ANYONE HELP.



___
NOCC, http://nocc.sourceforge.net




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Henry Devito]

BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1

Do you have a contact email for these guys? I couldn't see anything listed 
on their site anywhere. Seems the site is in current development.

Matt
Hi Matt,
I was just getting ready to try to order a IP phone and ATA in the morning. 
This is the contact info I have.

a.. email: [EMAIL PROTECTED]
a.. Phone: +49 69 949 44 185
a.. Fax: +49 69 949 44 118
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP USB Phone?

[david]

Hi,Adi,
We provide the USB phone you wanted, it can access Asterisk natively.  It 
can support Skype,X-Lite,X-PRO,eyeBeam,StanaPhone,SJphone,Net2Phone,Firefly and 
MSN too.  To get more information about that, contact with me offline or goto 
our website please.
Regards.

David at iaxtalk.com
http://www.iaxtalk.com


- Original Message - 
From: "Adi Linden" <[EMAIL PROTECTED]>
To: 
Sent: Monday, January 24, 2005 10:16 AM
Subject: [Asterisk-Users] SIP USB Phone? 


> There are a number of Skype USB phones available. Are there any when
> connected to a Windows PC can access Asterisk?
> 
> Adi
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP USB Phone?

[dean collins]

There are heaps but why not use a headset
If you insist on usb handset then there are 3 listed here.
http://www.telecoms.co.uk/catalog/default.php?cPath=583_829_830



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adi Linden
Sent: Sunday, January 23, 2005 9:16 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP USB Phone? 

There are a number of Skype USB phones available. Are there any when
connected to a Windows PC can access Asterisk?

Adi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP USB Phone?

[asterisk]

Adi,

Yes there are...

You can probably use the exact same skype USB phone with X-Lite or one of
the many other windows SIP softphones.
It is not a matter of being compatable with Asterisk so much as being
compatable with your Asterisk softphone..

In the X-Lite menu, system settings - USB Settings.

If you can spare the dollars, a hardware phone is almost always better
though..

Cheers
Shane 


> Adi Linden
> Sent: Monday, 24 January 2005 1:16 PM
> 
> There are a number of Skype USB phones available. Are there 
> any when connected to a Windows PC can access Asterisk?
> 
> Adi

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario?

[Sergey Kuznetsov]

You are very welcome!


All the Best!
Sergey.


Robert Augustyn wrote:

  
  
  Thanks for your help Sergey.
  robert
  
  
  From:
  [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
  On Behalf Of Sergey
Kuznetsov
  Sent: Sunday, January 23, 2005 8:00 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Can anyone recomentT1/PRI
providerin SouthOntario?
  
  
I got my PRIs from ISPtel as an add-on to my colo with MCI and thru
MCI. I'll try to find ISPtel web-site (if it's exists) thru
MCI's customer service. Actually Allstream's PRI will cost you around
700-750 CAD per month. It's not that bad.
  
I got just few PRIs with set of DIDs I need. This is enough for me. I
can set any ANI/C*ID form my range on my PRIs.
My incoming DNIS is 10-digit length.
I didn't try if I can port existing DIDs from another ILECs/CLECs.
  
  
All the Best!
Sergey.
  
  
Robert Augustyn wrote:
  

Thanks
You sure have to have experience
...:)
Do you know how I can contact
ISPtel?
Sprint quoted me a realy high
number.
btw: what do you get with your
PRI service?
robert
 


 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Sergey Kuznetsov
Sent: Sunday, January 23, 2005 5:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
providerin SouthOntario?


Sorry, I completely forgot. You have to have an experience how to use
the CRTC site =)
If you will click to "Public Proceedings" at the top of the main page
you will be redirected to
the page witch will show you the most of the useful information.
At that page in the "Telecommunications" Part of the table you will see
link "Tariff" with is
going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm

At that pages you have to choose year and then the name of the company
you are interesting about.
There is the some info buried there, but it's quite easy to find it.

I cannot find the website of ISPtel either. But I have the PRIs from
them and it's 2 times cheaper then PRIs from
Sprint.

http://www.crtc.gc.ca/8740/frn/2002/a4.htm
- Allstream (AT&T) rates.
Probably there is some new rates. Have to go thru all recent years.

All the Best!
Sergey.

Robert Augustyn wrote:

  
  Sergey,
  Thanks for the input.
  I looked at the crtc site did
few searches but I guess I do not know what to look for because I did
not find anything related to tariffs.
  On the same note I am not able
to find a Isptel web site either  I guess it is not my day today :)
  robert
   
  
  
   From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
  On Behalf Of Sergey Kuznetsov
  Sent: Sunday, January 23, 2005 4:15 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
provider in SouthOntario?
  
  
MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.
  
The best way to find rates for such lines to go to CRTC site and check
the tariffs for that.
  
  
All the Best!
Sergey.
  
  
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP USB Phone?

[Adi Linden]

There are a number of Skype USB phones available. Are there any when
connected to a Windows PC can access Asterisk?

Adi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Festival

[Matt Riddell]

Howard Lowndes wrote:
Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?
Is this a case of having to use AGI, or is there a simpler way?
Most people would use AGI for that (combined with the text2wave or 
whatever program).  In fact there may even be an example on the wiki.

--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can anyone recomentT1/PRI providerin SouthOntario?

[Robert Augustyn]

Thanks for your help Sergey.
robert


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey 
KuznetsovSent: Sunday, January 23, 2005 8:00 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Can anyone recomentT1/PRI providerin 
SouthOntario?
I got my PRIs from ISPtel as an add-on to my colo with MCI and thru 
MCI. I'll try to find ISPtel web-site (if it's exists) thruMCI's customer 
service. Actually Allstream's PRI will cost you around 700-750 CAD per month. 
It's not that bad.I got just few PRIs with set of DIDs I need. This is 
enough for me. I can set any ANI/C*ID form my range on my PRIs.My incoming 
DNIS is 10-digit length.I didn't try if I can port existing DIDs from 
another ILECs/CLECs.All the Best!Sergey.Robert 
Augustyn wrote: 

  
  Thanks
  You sure have to have experience 
...:)
  Do you know how I can contact ISPtel?
  Sprint quoted me a realy high number.
  btw: what do you get with your PRI 
  service?
  robert
   
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 
  5:54 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI 
  providerin SouthOntario?Sorry, I completely forgot. You 
  have to have an experience how to use the CRTC site =)If you will click to 
  "Public Proceedings" at the top of the main page you will be redirected 
  tothe page witch will show you the most of the useful information.At 
  that page in the "Telecommunications" Part of the table you will see link 
  "Tariff" with isgoing to this page: http://www.crtc.gc.ca/8740/eng/tariff.htmAt 
  that pages you have to choose year and then the name of the company you are 
  interesting about.There is the some info buried there, but it's quite easy 
  to find it.I cannot find the website of ISPtel either. But I have the 
  PRIs from them and it's 2 times cheaper then PRIs fromSprint.http://www.crtc.gc.ca/8740/frn/2002/a4.htm 
  - Allstream (AT&T) rates.Probably there is some new rates. Have to go 
  thru all recent years.All the Best!Sergey.Robert Augustyn 
  wrote: 
  

Sergey,
Thanks for the input.
I looked at the crtc site did few searches but I guess 
I do not know what to look for because I did not find anything related to 
tariffs.
On the same note I am not able to find a Isptel web 
site either  I guess it is not my day today :)
robert
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] 
On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 
2005 4:15 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment 
T1/PRI provider in SouthOntario?MCI does not provide 
voice trunks T1/PRI by itself. They resell it as a add-ons to their IP 
solutions.Sprint is expensive. Bell is quite expensive as well. 
Allstream quite better in price. ISPTel is the least expensiveone but 
their customer support is not one of the best.The best way to find 
rates for such lines to go to CRTC site and check the tariffs for 
that.All the Best!Sergey.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] - New Security List -

[steve szmidt]

I wanted to let everyone know that we have a new security list covering, 
Security, of all things.

You can sign up at:
 http://lists.digium.com/mailman/listinfo/asterisk-security

The idea here is to have a less busy list for security issues, as the -users 
is hard to keep up with already. Fortunately there's already data known about 
security and Asterisk. My interest is in being able to stay on top of issues 
that may show up and quickly get it solved as a team effort.

By pulling resources we can test and verify a lot faster than doing it alone. 
Best practices can be assembled, tested and posted to the wiki for all.

No doubt this will be a low volume list, but you are all welcome!

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Matt Gibson]

Henry Devito wrote:
BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1

Do you have a contact email for these guys? I couldn't see anything 
listed on their site anywhere. Seems the site is in current development.

Matt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Upgrade to the newest cvs now asterisk will notstart

[Eric Hall]

Looking at the error I tried moving chan_modem* out of the modules
folder and asterisk started and its working again...


Not sure what changed in the chan_modem_i4l.so but removing it from the
folder fixed my problem.





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Hall
Sent: Sunday, January 23, 2005 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Upgrade to the newest cvs now asterisk will
notstart

Hello group
 I just update to the newest CVS now I'm not able to get asterisk to
start. No error during the make or make install


I did a make clean before the make;make install

Any help would be great


Here is the output

asterisk -vgcd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf
  == Binding realtime_ext to mysql/realtime/extensions_table
  == Binding voicemail to mysql/realtime/voicemail_users
  == Binding sipfriends to mysql/realtime/sip_buddies Asterisk
CVS-HEAD-01/23/05-19:38:48, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

=
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands Asterisk Management
interface listening on port 5038
  == RTP Allocating from port range 1 -> 2 Asterisk PBX Core
Initializing Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [ImportVar]
  == Registered application 'ImportVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
 [chan_modem.so] => (Generic Voice Modem Driver)  [res_musiconhold.so]
=> (Music On Hold Resource)
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
Junk at the beginning 49443302
Warning, flexibel rate not heavily tested!
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!
 [res_adsi.so] => (ADSI Resource)
 [res_features.so] => (Call Parking Resource)
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls  [res_crypto.so] =>
(Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
 [res_indications.so] => (Indications Configuration)
-- Registered indication country 'cl'
-- Registered indication country 'tw'
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Registered indication country 'za'
-- Registered indication country 'it'
-- Registered indication country 'us-o'
-- Registered indication 

[Asterisk-Users] VoIP software for MAC OS older than "X"?

[Daiku]

Hello,

is there anybody reading this who has experience with VoIP (IAX or not) on
Macintosh computers? If so, have you ever seen or heard of (even an
experimental, i.e., never marketed) VoIP application for any of the older
Mac OSs, such as 9, 8, or 7?

I can't quite believe that VoIP is such a recent idea that it was invented
only *after* Mac OS X had become firmly established, but so far my searches
have turned out nothing. However, not all good stuff and good ideas are on
the web,so a community of knowledgable people often has information that a
web search cannot produce.

Appreciate any leads and comments...

Thanks: H.D.

--



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Henry Devito]

BTW they also an iax2 ATA
Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message - 
From: "Erik Espinoza" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Sunday, January 23, 2005 6:40 PM
Subject: Re: [Asterisk-Users] IAXy's apparantly failing in the field


I have one of these phones. I bought it off of eBay. Not sure where to
get them direct. You will need to load the proper image, in that I
believe it ships with SIP by default. Each protocol has its own image.
Erik
On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
Michael Giagnocavo wrote:
>>Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
>>only game in town. I know that the Farfon device will be out soon and
>>we'll be able to judge its quality at that time.
>>
>>
>
>Or any PA168 phones, which are already out, and support IAX2, SIP, 
>H323,
>MGCP and N2P. (I've got one on my desk here as do a few others, and it 
>works
>great.)
>
>
I want one of them!

Which model is it? Did you have to do any software upgrade? How much
does it cost?
Cheers,
Jean-Michel.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] VoIP Providers and Backbone Servers

[Subhi S Hashwa]

Monday, January 24, 2005, 12:20:46 AM, Jay Milk wrote:

> I don't want to be a kill-joy, but after reading your various messages
> over the last few days, I think you're in over your head on this one.  I
> suggest you first get your own * system up and running.  Then,
> re-examine your goals.  So far, you don't seem to be adding anything new
> to the VOIP community, so I'm at a loss at how you expect to make money.
> If you're reselling services, then you can't compete with the current
> players in the market, as obviously their prices would be better than
> yours.  If you're trying to bundle resold services, the big ones (like
> Vonage) have an insurmountable advantage in infrastructure, volume and
> installed base.  And lastly, if you expect to sell service along with
> your own softphone client (which you still have to complete), there are
> free solutions with established services out there (iconnecthere, etc).

> So, besides (weak) competition, what are you going to add?

Competition is what drives prices down, VoIP is a new challange to
traditional telecoms many are starting to wakeup to the fact that soon
you will no longer have an area code, country code, but rather a
global number that is yours and will follow you where you go globally,
a much bigger version of the GSM roaming we had in europe for years
without all the extra silly costs. once you have IP you have a phone,
you can collect messages, phone people and do your work on the move.

With the explosion of wireless hotspots this makes a VoIP on the move
a reality, some telecos are now realising that if they don't join the
'revolution' they'll be out of profitable call business and will endup
installing circuits for other voip providers to use.

BT for example here in the UK are planning to convert their entire
network to VoIP based network. IP to everyhome and that sort of thing
because they want a slice of the action. My guess is telecos will move
away from charging for calls and start charging for VoIP traffic but
it is a rapid development and my guess is some telecos will be caught
out and probably be out of business in the next 5-10 years.

Competition however small encourages companies to improve their
services, compete on pricing and look for ways to attract new
customers, many companies I know now use skype to contact staff
working from home, because it is convenient and easy to use.

Like I said, compeition isn't a bad thing, some people work on * for
the sake of technology development and some work on it in the hope of
making a living out of it.

Just my £0.02 :)


-- 
Best regards,
 Subhi S Hashwamailto:[EMAIL PROTECTED]
 When everything is heading your way, you're in the wrong lane.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Henry Devito]

Try here for the iax2 phone
http://www.ngtel.de/products.php#1
- Original Message - 
From: "Erik Espinoza" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Sunday, January 23, 2005 6:40 PM
Subject: Re: [Asterisk-Users] IAXy's apparantly failing in the field


I have one of these phones. I bought it off of eBay. Not sure where to
get them direct. You will need to load the proper image, in that I
believe it ships with SIP by default. Each protocol has its own image.
Erik
On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
Michael Giagnocavo wrote:
>>Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
>>only game in town. I know that the Farfon device will be out soon and
>>we'll be able to judge its quality at that time.
>>
>>
>
>Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
>MGCP and N2P. (I've got one on my desk here as do a few others, and it 
>works
>great.)
>
>
I want one of them!

Which model is it? Did you have to do any software upgrade? How much
does it cost?
Cheers,
Jean-Michel.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Upgrade to the newest cvs now asterisk will not start

[Eric Hall]

Hello group
 I just update to the newest CVS now I'm not able to get asterisk to
start. No error during the make or make install


I did a make clean before the make;make install

Any help would be great


Here is the output

asterisk -vgcd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf
  == Binding realtime_ext to mysql/realtime/extensions_table
  == Binding voicemail to mysql/realtime/voicemail_users
  == Binding sipfriends to mysql/realtime/sip_buddies
Asterisk CVS-HEAD-01/23/05-19:38:48, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>

=
Asterisk Event Logger Started /var/log/asterisk/event_log
  == Manager registered action Ping
  == Manager registered action Events
  == Manager registered action Logoff
  == Manager registered action Hangup
  == Manager registered action Status
  == Manager registered action Setvar
  == Manager registered action Getvar
  == Manager registered action Redirect
  == Manager registered action Originate
  == Manager registered action MailboxStatus
  == Manager registered action Command
  == Manager registered action ExtensionState
  == Manager registered action AbsoluteTimeout
  == Manager registered action MailboxCount
  == Manager registered action ListCommands
Asterisk Management interface listening on port 5038
  == RTP Allocating from port range 1 -> 2
Asterisk PBX Core Initializing
Registering builtin applications:
 [AbsoluteTimeout]
  == Registered application 'AbsoluteTimeout'
 [Answer]
  == Registered application 'Answer'
 [BackGround]
  == Registered application 'BackGround'
 [Busy]
  == Registered application 'Busy'
 [Congestion]
  == Registered application 'Congestion'
 [DigitTimeout]
  == Registered application 'DigitTimeout'
 [Goto]
  == Registered application 'Goto'
 [GotoIf]
  == Registered application 'GotoIf'
 [GotoIfTime]
  == Registered application 'GotoIfTime'
 [Hangup]
  == Registered application 'Hangup'
 [NoOp]
  == Registered application 'NoOp'
 [Prefix]
  == Registered application 'Prefix'
 [Progress]
  == Registered application 'Progress'
 [ResetCDR]
  == Registered application 'ResetCDR'
 [ResponseTimeout]
  == Registered application 'ResponseTimeout'
 [Ringing]
  == Registered application 'Ringing'
 [SayNumber]
  == Registered application 'SayNumber'
 [SayDigits]
  == Registered application 'SayDigits'
 [SayAlpha]
  == Registered application 'SayAlpha'
 [SayPhonetic]
  == Registered application 'SayPhonetic'
 [SetAccount]
  == Registered application 'SetAccount'
 [SetAMAFlags]
  == Registered application 'SetAMAFlags'
 [SetGlobalVar]
  == Registered application 'SetGlobalVar'
 [SetLanguage]
  == Registered application 'SetLanguage'
 [SetVar]
  == Registered application 'SetVar'
 [ImportVar]
  == Registered application 'ImportVar'
 [StripMSD]
  == Registered application 'StripMSD'
 [Suffix]
  == Registered application 'Suffix'
 [Wait]
  == Registered application 'Wait'
 [WaitExten]
  == Registered application 'WaitExten'
Asterisk Dynamic Loader Starting:
 [chan_modem.so] => (Generic Voice Modem Driver)
 [res_musiconhold.so] => (Music On Hold Resource)
  == Registered application 'MusicOnHold'
  == Registered application 'WaitMusicOnHold'
  == Registered application 'SetMusicOnHold'
Junk at the beginning 49443302
Warning, flexibel rate not heavily tested!
Junk at the beginning 49443303
Warning, flexibel rate not heavily tested!
 [res_adsi.so] => (ADSI Resource)
 [res_features.so] => (Call Parking Resource)
  == Registered application 'ParkedCall'
  == Registered application 'Park'
  == Manager registered action ParkedCalls
 [res_crypto.so] => (Cryptographic Digital Signatures)
-- Loaded PUBLIC key 'iaxtel'
-- Loaded PUBLIC key 'freeworlddialup'
 [res_indications.so] => (Indications Configuration)
-- Registered indication country 'cl'
-- Registered indication country 'tw'
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Registered indication country 'za'
-- Registered indication country 'it'
-- Registered indication country 'us-o'
-- Registered indication country 'gr'
-- Registered indication country 'ru'
-- Registered indication country 'nz'
-- Setting default indication country to 'us'
  == Registered application 'Playtones'
  == Registered application 'StopPlaytones'
 [res_monitor.so] => (Call Monitoring Resource)
  == Registered application 'Monitor'
  == Registered application 'StopMonitor'
  == Registered application 'ChangeMonitor'
  == Manager registered action Monitor
  == Manager registered action StopMonitor
  == Manager regis

Re: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?

[Sergey Kuznetsov]

I got my PRIs from ISPtel as an add-on to my colo with MCI and thru
MCI. I'll try to find ISPtel web-site (if it's exists) thru
MCI's customer service. Actually Allstream's PRI will cost you around
700-750 CAD per month. It's not that bad.

I got just few PRIs with set of DIDs I need. This is enough for me. I
can set any ANI/C*ID form my range on my PRIs.
My incoming DNIS is 10-digit length.
I didn't try if I can port existing DIDs from another ILECs/CLECs.


All the Best!
Sergey.


Robert Augustyn wrote:

  
  
  Thanks
  You sure have to have experience
...:)
  Do you know how I can contact
ISPtel?
  Sprint quoted me a realy high
number.
  btw: what do you get with your
PRI service?
  robert
   
  
  
  From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Sergey
Kuznetsov
  Sent: Sunday, January 23, 2005 5:54 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
providerin SouthOntario?
  
  
Sorry, I completely forgot. You have to have an experience how to use
the CRTC site =)
If you will click to "Public Proceedings" at the top of the main page
you will be redirected to
the page witch will show you the most of the useful information.
At that page in the "Telecommunications" Part of the table you will see
link "Tariff" with is
going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm
  
At that pages you have to choose year and then the name of the company
you are interesting about.
There is the some info buried there, but it's quite easy to find it.
  
I cannot find the website of ISPtel either. But I have the PRIs from
them and it's 2 times cheaper then PRIs from
Sprint.
  
  http://www.crtc.gc.ca/8740/frn/2002/a4.htm
- Allstream (AT&T) rates.
Probably there is some new rates. Have to go thru all recent years.
  
All the Best!
Sergey.
  
Robert Augustyn wrote:
  

Sergey,
Thanks for the input.
I looked at the crtc site did
few searches but I guess I do not know what to look for because I did
not find anything related to tariffs.
On the same note I am not able
to find a Isptel web site either  I guess it is not my day today :)
robert
 


 From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
On Behalf Of Sergey Kuznetsov
Sent: Sunday, January 23, 2005 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
provider in SouthOntario?


MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.

The best way to find rates for such lines to go to CRTC site and check
the tariffs for that.


All the Best!
Sergey.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  
  
  

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] No music with "Blind" transfer on GS ATA + Sipura-841

[Justin Moore]

Hi there,
I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream 
ATA's.

The problem is that with both of these devices the Unattended call 
transfer process seems to be just like Attended but instead you hang up 
as soon as you have dialled the number of the party your are 
transferring to.  The call transfer all works fine BUT as you complete 
your side of the transfer and the destination extension is ringing - ie. 
Caller calls you, you transfer call to another extension and hangup 
before they answer and another extension is left ringing waiting for 
someone to pickup, the inital caller has only silence and no MOH (or 
ringing).

I cannot tell wether this is a function of Asterisk (which seems to end 
MOH during bridging of the calls) or the SPA-841 / GS's but the problem 
is that doing this type of unattended transfer results in what the 
initial caller hears as a dead line and they are prone to hangup should 
the called party not answer quickly. 

I guess we could just do all transfers as attended and wait for the 
called party to answer so as to avoid this "assumed dead line" problem 
but I wonder if anyone has also experienced this and or found a solution.

My Asterisk log is as follows:
asterisk*CLI>
   -- Executing Macro("SIP/201-6447", "oneline|Sip/200") in new stack
   -- Executing SetMusicOnHold("SIP/201-6447", "random") in new stack
   -- Executing Dial("SIP/201-6447", "Sip/200|30|tr") in new stack
   -- Called 200
   -- SIP/200-08b4 is ringing
   -- SIP/200-08b4 answered SIP/201-6447
   -- Attempting native bridge of SIP/201-6447 and SIP/200-08b4
   -- Started music on hold, class 'random', on SIP/201-6447
   -- Executing Macro("SIP/200-eedc", "oneline|Sip/202") in new stack
   -- Executing SetMusicOnHold("SIP/200-eedc", "random") in new stack
   -- Executing Dial("SIP/200-eedc", "Sip/202|30|tr") in new stack
   -- Called 202
   -- SIP/202-f8cf is ringing
   -- Stopped music on hold on SIP/201-6447 
 == Spawn extension (macro-oneline, s, 2) exited non-zero on 
'SIP/200-eedc' in macro 'oneline'
 == Spawn extension (from-internal, 200, 1) exited non-zero on 
'SIP/200-eedc'
   -- SIP/202-f8cf answered SIP/201-6447
   -- Attempting native bridge of SIP/201-6447 and SIP/202-f8cf

Thanks,
Justin.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] VoIP Providers and Backbone Servers

[lonnie]

I really appreciate your comments regarding the challenges that face a new
VoIP service. While it is true that there is always still much to learn in
both the VoIP arena and with a true business model that will try to bring
something new and exciting to the existing community, I still strongly
contend that as this is still a young industry that building
conglomerations of existing VoIP services along with strategic partnering
will foster industry innovation and lay a strong foundation for the VoIP
industry to eventually overtake the analog corporate strong-holds that
currently exist.

Additionally, these small beginings enable people like myself to learn the
industry quickly and get involved. It also allow us to learn about the
Astrisk PBX system as well as the multitude of hardware and software that
comprise this exciting field.

In as much a funding is concerned, we intend to come online just as in the
manner as other VoIP services and to move quickly to branch out by
partnering to cover a broader scoping topology. Could be challenging at
first, but your response implies that an Internet store could not come
online just because there are other currently existing online stores in
that market. It's a long hard road, but I do believe that in order to
become "big" you must plan from the onset to be "big" and think in those
terms. Everyday should be the question in your mind of "How can I expand
the business today?". Always this and always moving forward; never back.

With regards to the softphone, once finished, it would be given away for
free and probably open source as well.

We are here to promote the industry, make it grow, and build a solid
foundation for the future of telecommunications. At least that is what I
see in this. Hope that it is the same for you as well.

Just my personal input to the email that you sent to the list and I am
sure that there will be some disagreement with its content and to the
philosophy indicated but this is an exciting time for getting started and
to embark upon the multitude of innovation that awaits us all.

I truly thank all of the members of the list for giving me an opportunity
to learn from you and to, hopefully, eventually be able to give back to
the list and help others in the near future. There is so much that the
Astrisk PBX and supporting hardware can do to open doors on the virtal
highway and I am happy to begin travelling down that road.

Thanks again for your response and have a great day,
Lonnie Cumberland

> I don't want to be a kill-joy, but after reading your various messages
> over the last few days, I think you're in over your head on this one.  I
> suggest you first get your own * system up and running.  Then,
> re-examine your goals.  So far, you don't seem to be adding anything new
> to the VOIP community, so I'm at a loss at how you expect to make money.
> If you're reselling services, then you can't compete with the current
> players in the market, as obviously their prices would be better than
> yours.  If you're trying to bundle resold services, the big ones (like
> Vonage) have an insurmountable advantage in infrastructure, volume and
> installed base.  And lastly, if you expect to sell service along with
> your own softphone client (which you still have to complete), there are
> free solutions with established services out there (iconnecthere, etc).
>
> So, besides (weak) competition, what are you going to add?
>
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
>> Sent: Sunday, January 23, 2005 2:59 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [Asterisk-Users] VoIP Providers and Backbone Servers
>>
>>
>> Hello All,
>>
>> Well, my explorations in to the world of VoIP is proving
>> fruitful and in the near future I am hoping to have my small
>> VoIP online service up and running ready to help promote the
>> industry and hopefully gain a few customers in the process.
>>
>> Additionally, I will soon have my IAX and SIP softphone ready
>> that will handle video, audio, and text communications.
>>
>> I am looking for quality and fair priced service providers so
>> that I can add some of thier servies to my VoIP service which
>> will start with an Asterisk PBX and some reliable Billing
>> software (still trying to decide, but Trebas or ASTPP looks
>> like it will work for me to get started.)
>>
>> In particular I will be providing Phone-Phone, Phone-PC,
>> PC-PC, and PC-Phone connections.
>>
>> I am looking for services like
>>
>> 1. PSTN Termination Services (Good International Rates)
>> 2. 800 Tollfree  access line services
>> 3. local, national, and international analog access line
>> services in addition to my Asterisk PBX, if they exist (for
>> my Phone-* services)
>>
>> and other services that you think are useful.
>>
>> Any suggestions or comments are appreciated and if you know
>> of a quality service that I am looking for then now is the
>> time and I invite your responses to this email.
>>
>> Thanks

[Asterisk-Users] Data calls with Asterisk

[Karim Mardhani]

  Hi All:  I am new to Asterisk so if my question sounds too 
  newbeeish then pleasebear with me.  I have about 10 remote 
  locations which are collecting some data.  Iwould like to upload that 
  data every night.  All remote locations have56K modem.  I was 
  wondering can Asterisk be used to receive this data? Basically I will have 
  an asterisk with 1 FXO card and have it receivedata calls.  Can 
  asterisk receive data calls?  Thanks in advance for your 
  responsesRegards,Karim MardhaniZeeCore 
Consulting
   

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Erik Espinoza]

I have one of these phones. I bought it off of eBay. Not sure where to
get them direct. You will need to load the proper image, in that I
believe it ships with SIP by default. Each protocol has its own image.

Erik


On Mon, 24 Jan 2005 00:54:48 +0400, Jean-Michel Hiver
<[EMAIL PROTECTED]> wrote:
> Michael Giagnocavo wrote:
> 
> >>Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
> >>only game in town. I know that the Farfon device will be out soon and
> >>we'll be able to judge its quality at that time.
> >>
> >>
> >
> >Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
> >MGCP and N2P. (I've got one on my desk here as do a few others, and it works
> >great.)
> >
> >
> I want one of them!
> 
> Which model is it? Did you have to do any software upgrade? How much
> does it cost?
> 
> Cheers,
> Jean-Michel.
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Definity PBX with a T100P & TN767E

[Doug Lytle]

Ken Godee wrote:

I'm currently playing with a Digium T100P card and 2 Grandstream 
phones, things are working well.  I wanted to move on to linking our 
Definity G3R Rev 8.2 to the T100P.  Everything that I've read so far 
shows that you need a TN464 to accomplish this.  We have a TN767E 
available.

Inbound/outbound, DID from G3 inbound, ext./ext., etc.

Ken,
Hope your holiday went well.
We have been able to get our Definity G3R working with Asterisk via a 
T100P card and a TN767E card, works very well!  But, I'm a little stuck 
on how to get the DID info from the G3 and ext/ext info to the G3.  
Incoming shows the trunk info setup by our phone admin.

Happen to have a link that you could point me to on this setup?
Thanks again!
Doug Lytle
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] VoIP Providers and Backbone Servers

[Jay Milk]

I don't want to be a kill-joy, but after reading your various messages
over the last few days, I think you're in over your head on this one.  I
suggest you first get your own * system up and running.  Then,
re-examine your goals.  So far, you don't seem to be adding anything new
to the VOIP community, so I'm at a loss at how you expect to make money.
If you're reselling services, then you can't compete with the current
players in the market, as obviously their prices would be better than
yours.  If you're trying to bundle resold services, the big ones (like
Vonage) have an insurmountable advantage in infrastructure, volume and
installed base.  And lastly, if you expect to sell service along with
your own softphone client (which you still have to complete), there are
free solutions with established services out there (iconnecthere, etc).

So, besides (weak) competition, what are you going to add?

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
> Sent: Sunday, January 23, 2005 2:59 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] VoIP Providers and Backbone Servers
> 
> 
> Hello All,
> 
> Well, my explorations in to the world of VoIP is proving 
> fruitful and in the near future I am hoping to have my small 
> VoIP online service up and running ready to help promote the 
> industry and hopefully gain a few customers in the process.
> 
> Additionally, I will soon have my IAX and SIP softphone ready 
> that will handle video, audio, and text communications.
> 
> I am looking for quality and fair priced service providers so 
> that I can add some of thier servies to my VoIP service which 
> will start with an Asterisk PBX and some reliable Billing 
> software (still trying to decide, but Trebas or ASTPP looks 
> like it will work for me to get started.)
> 
> In particular I will be providing Phone-Phone, Phone-PC, 
> PC-PC, and PC-Phone connections.
> 
> I am looking for services like
> 
> 1. PSTN Termination Services (Good International Rates)
> 2. 800 Tollfree  access line services
> 3. local, national, and international analog access line 
> services in addition to my Asterisk PBX, if they exist (for 
> my Phone-* services)
> 
> and other services that you think are useful.
> 
> Any suggestions or comments are appreciated and if you know 
> of a quality service that I am looking for then now is the 
> time and I invite your responses to this email.
> 
> Thanks to everyone on this for giving me such great help, Lonnie
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/aster> isk-users
> To 
> UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Music On-Hold problem

[C F]

What is the CLI output you are getting?
Do you have a timer source installed?



On Sun, 23 Jan 2005 18:10:27 -0500, Computer Onsite Support
<[EMAIL PROTECTED]> wrote:
> My problem is: No matter what machine I install and configure Asterisk on it
> I just can't get the music on-hold to work. Is anyone of you out there have
> such problem? If so what have you done to fix the problem? I've tried so far
> three other computers and none of them I was able to get music on-hold to
> work.
> 
> CAN PLEASE ANYONE HELP.
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can anyone recoment T1/PRI providerin SouthOntario?

[Robert Augustyn]

Thanks
You sure have to have experience ...:)
Do you know how I can contact ISPtel?
Sprint quoted me a realy high number.
btw: what do you get with your PRI 
service?
robert
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey 
KuznetsovSent: Sunday, January 23, 2005 5:54 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Can anyone recoment T1/PRI providerin 
SouthOntario?
Sorry, I completely forgot. You have to have an experience how to use 
the CRTC site =)If you will click to "Public Proceedings" at the top of the 
main page you will be redirected tothe page witch will show you the most of 
the useful information.At that page in the "Telecommunications" Part of the 
table you will see link "Tariff" with isgoing to this page: http://www.crtc.gc.ca/8740/eng/tariff.htmAt 
that pages you have to choose year and then the name of the company you are 
interesting about.There is the some info buried there, but it's quite easy 
to find it.I cannot find the website of ISPtel either. But I have the 
PRIs from them and it's 2 times cheaper then PRIs fromSprint.http://www.crtc.gc.ca/8740/frn/2002/a4.htm 
- Allstream (AT&T) rates.Probably there is some new rates. Have to go 
thru all recent years.All the Best!Sergey.Robert Augustyn 
wrote: 

  
  Sergey,
  Thanks for the input.
  I looked at the crtc site did few searches but I guess I 
  do not know what to look for because I did not find anything related to 
  tariffs.
  On the same note I am not able to find a Isptel web site 
  either  I guess it is not my day today :)
  robert
   
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Sergey KuznetsovSent: Sunday, January 23, 2005 
  4:15 PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [Asterisk-Users] Can anyone recoment T1/PRI 
  provider in SouthOntario?MCI does not provide voice 
  trunks T1/PRI by itself. They resell it as a add-ons to their IP 
  solutions.Sprint is expensive. Bell is quite expensive as well. Allstream 
  quite better in price. ISPTel is the least expensiveone but their customer 
  support is not one of the best.The best way to find rates for such 
  lines to go to CRTC site and check the tariffs for that.All the 
  Best!Sergey.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: Best VPN server for * and woad warriors using windows?

[Calvin Hendryx-Parker]

Tom Ivar Helbekkmo wrote:
Me too, and I'd certainly use it in the original poster's stead.
However, he specifically said that he must have an IPSEC tool, and
OpenVPN is not IPSEC.
-tih
 

We are currently using OpenVPN too with good success.  I'm not sure why 
you would require IPSEC.  I thought that is what we wanted when we 
looked at VPN solutions, but OpenVPN is so easy compared the the IPSEC 
stuff out there.

Calvin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Festival

[Howard Lowndes]

Is it possible to get the Festival command to read the text from a
system file rather than having it input as a text string?

I suppose I could put the text string into an Asterisk variable and
reference that in the Festival command, but then, how do I get the
contents of the file into the Asterisk variable?

Is this a case of having to use AGI, or is there a simpler way?

-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Sip Notify and PHP AGI

[Krystian Filiks]

Hello * Users.

 

I need to be able to generate a Sip Notify message using PHP
AGI but have no idea how I can do that.

 

What I need to send is the balance of the prepaid card and
display it on the soft phones display.

 

Does anyone know how to do this?

 

Thanks in advance.

KF






___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SPA-2000

[Chris Stenton]

600 is for the US only.
FXS impedence for
UK  370+620||310nF
Europe CTR21 270+750||150nF
Chris
- Original Message - 
From: "Remco Barende" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Saturday, January 22, 2005 10:12 AM
Subject: Re: [Asterisk-Users] SPA-2000


On Sat, 22 Jan 2005, Duane wrote:
Remco Barende wrote:
On Fri, 21 Jan 2005, Henry Devito wrote:
Hi,  I have not implemented any of the spa-2000's yet.  Do they work ok 
with asterisk?  Is the 2000 capable of having 2 FXS extensions off each 
one or is it two fxs ports with the same extension?

They work pretty well, but I'm not impressed with the sound quality. 
Sound is quite soft and I have to adjust the input and output gains to 
something like +3 or +5 for in+out and then an annoying hiss is audible.
I have a sipura 2000 and haven't had to alter gain at all, and no hiss, 
then again are you using ulaw or using g729?
I use G711u. When I do not adjust the output gain the volume in+out is 
just too soft. Or would I need to change another setting?  Under Regional 
I can also set port impedance. No idea though if the default value of 600 
is ok for Europe?

As for the original question, the 2 ports on the 2000 and the 3000 are 
both seperate SIP identities and you have to configure them as 2 seperate 
lines...


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Music On-Hold problem

[Computer Onsite Support]

My problem is: No matter what machine I install and configure Asterisk on it
I just can't get the music on-hold to work. Is anyone of you out there have
such problem? If so what have you done to fix the problem? I've tried so far
three other computers and none of them I was able to get music on-hold to
work.

CAN PLEASE ANYONE HELP.
<>___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Autio cut off at beginning of call

[Adi Linden]

My wife brought to my attention just yesterday that this is happening on
all my inbound PSTN calls. I am using a ZAP interface, not IAX.

Adi
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] T1 Timing Slips

[Ken Godee]

Ken Godee wrote:
Does anyone know how to monitor * to see if they are receiving timing 
slips
on a span connected to a T100P card?  I am seeing b-channel restarts 
quite
often and also getting "No D-channels available" warnings from time to 
time.
Yesterday I had all the b-channels crash during a MeetMe Conference.  Not
good!  This PRI is connected to an Avaya Definity PBX that is onsite and
located in the same room as *.  * is set to clock off the Definity.   
I am
seeing no problems on the PRI from the Definity side.

Just thought I'd run this by you.
We've been running connected to our Definity G3si R6
via TN767 <--> TE410P and have had no problems.
I guess I'll eat crow alittle bit
I guess your email made me focus a little more into it.
I'm also having the same problems as you are, D-Channel bouncing.
D-channel down and right back up and then b-channels restart, while
restarting they "DO" drop any active channels. :(
Experenced first hand on friday while remote monitoring and
on a call.
D-channel down
No D-channel found, using channel 48 anyway.
D-channel up
restarting channel etc.
As another poster suggested, I tried changing timing to internal
clocking, vs. Definty, no help thou. I've done a ton of searching and 
have not found much more I can try.

What protocol are you using on the Definity side?
As I understand it "a = ni1 / b = national"
If you come across anything that helps, please let me know.
I'll also let you know if I find anything.
I also see no problems on the Definity side.
No errors when loop up circuit either.
ztmonitor runs 100%-99%.
No missing interrupts, etc.
Load/no load doesn't seem to make a difference.
Running astersk v1.0.3
ken


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?

[Sergey Kuznetsov]

Sorry, I completely forgot. You have to have an experience how to use
the CRTC site =)
If you will click to "Public Proceedings" at the top of the main page
you will be redirected to
the page witch will show you the most of the useful information.
At that page in the "Telecommunications" Part of the table you will see
link "Tariff" with is
going to this page: http://www.crtc.gc.ca/8740/eng/tariff.htm

At that pages you have to choose year and then the name of the company
you are interesting about.
There is the some info buried there, but it's quite easy to find it.

I cannot find the website of ISPtel either. But I have the PRIs from
them and it's 2 times cheaper then PRIs from
Sprint.

http://www.crtc.gc.ca/8740/frn/2002/a4.htm
- Allstream (AT&T) rates.
Probably there is some new rates. Have to go thru all recent years.

All the Best!
Sergey.

Robert Augustyn wrote:

  
  
  Sergey,
  Thanks for the input.
  I looked at the crtc site did
few searches but I guess I do not know what to look for because I did
not find anything related to tariffs.
  On the same note I am not able
to find a Isptel web site either  I guess it is not my day today :)
  robert
   
  
  
  From:
  [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]
  On Behalf Of Sergey
Kuznetsov
  Sent: Sunday, January 23, 2005 4:15 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Can anyone recoment T1/PRI
provider in SouthOntario?
  
  
MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.
  
The best way to find rates for such lines to go to CRTC site and check
the tariffs for that.
  
  
All the Best!
Sergey.
  
  
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Autio cut off at beginning of call

[Andrew Kohlsmith]

On January 23, 2005 03:42 pm, Reid Forrest wrote:
> When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
> VoicePulse Connect) I often find that after the call is answered the first
> few seconds of audio are cut off (i.e. I don't hear the called party). This
> usually results in the called party saying "hello Hello???" until I
> hear them.

This is not normal; I do *not* have this issue with NuFone and I have placed a 
ton of calls through them daily for the past year.  I don't recall having 
this problem with voicepulse connect when I used them, nor do I have the 
issue with iax.cc for inbound calls.

> Has anyone else experienced this problem and found a cause or fix? My
> internal calls are perfect. It's just Internet-terminated calls that have
> the problem. Someone wrote in response to the last post saying that the
> audio path probably wasn't set up yet. I think this is the symptom, but I'm
> wondering what's the cause, and if there's a fix.

It very much sounds like it's something on your end...  How about some 
specifics?

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Can anyone recoment T1/PRI provider in SouthOntario?

[Robert Augustyn]

Sergey,
Thanks for the input.
I looked at the crtc site did few searches but I guess I do 
not know what to look for because I did not find anything related to 
tariffs.
On the same note I am not able to find a Isptel web site 
either  I guess it is not my day today :)
robert
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Sergey 
KuznetsovSent: Sunday, January 23, 2005 4:15 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Can anyone recoment T1/PRI provider in 
SouthOntario?
MCI does not provide voice trunks T1/PRI by itself. They resell it as 
a add-ons to their IP solutions.Sprint is expensive. Bell is quite expensive 
as well. Allstream quite better in price. ISPTel is the least expensiveone 
but their customer support is not one of the best.The best way to find 
rates for such lines to go to CRTC site and check the tariffs for 
that.All the Best!Sergey.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] simulating multiple lines using ADSI

[Jon Gabrielson]

Does anyone have any experience with making an
adsi phone appear to have more than one line.
It seems like this would be a very simple and very useful
thing to be able to do.  Ideally, it would be nice if you
could make the 6 soft buttons appear as lines 1-6 and
if you press one of the soft buttons, it puts the current
line on hold and gives you a new dialtone.

Has anyone either done something like this or would
happen to know how to do something like this?


Thanks,


Jon.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Andrew Kohlsmith]

On January 23, 2005 04:04 pm, Mike Sander wrote:
> Is the harddisk activity on a standard asterisk install such that I don't
> really have to worry if the power cuts??

Not typically; there isn't much writing going on, this is true.  Are you that 
cash strapped that a $75 UPS with a serial port is out of your budget?

> As I understand, if HD activity is minimal, the probability of HD failure
> is significantly reduced.

HDDs don't fail because they lose power.  You get data corruption when writing 
and losing power, and you get filesystem corruption if the filesystem/OS is 
postponing writes to increase write performance.  That's not HDD failure.

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Some issues with X-Lite and codecs.

[Sergey Kuznetsov]

Yes I did. The same. It looks like there is some packet loss on the way 
to my VoIP box.
Is there any optimal settings for jitter buffer for * ?

All the Best!
Sergey.
Andrew Yager wrote:
Hi Sergey,
Have you tried phoning from X-Lite to your PSTN line, or your PSTN 
line to X-Lite? How is the audio quality then? Does it vary depending 
on the codec you have used?

Andrew
On 23/01/2005, at 4:31 PM, Sergey Kuznetsov wrote:
Hi there,
I am experiencing some issue with X-Lite.
When I am calling over the phone thru my PSTN-to-VoIP gateway 
internationally using G.729 the quality is just perfect.
When I am using X-Lite to connect the same box, and then to call 
internationally - I am experiencing some issues.
I have 5Mbit/800Kbit cable link with average 60 msecs to my VoIP box. 
The transfer rate is never falling below 500Kbytes/sec.
Therefore I am not suspecting quite noticeable packet loss.
I enabled G.711 ulaw, alaw and speex codecs on both sides. By playing 
with different codecs I am trying to avoid some
clicking and sound distortion, which is I am experiencing right now. 
Speex sometimes is better than G.711, but still having the same
glitching. My question is, is there any way to fix it by playing with 
some parameters on * side, or it's better to play with X-Lite 
parameters?


All the Best!
Sergey.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] UPS for Asterisk

[Jon Radon]

Why risk it?  Just go snag a cheap UPS from your local store.  Just
get something with enough run time to shut the system down gracefully.


On Mon, 24 Jan 2005 08:04:36 +1100, Mike Sander
<[EMAIL PROTECTED]> wrote:
> I'd considering an UPS backup system for my Asterisk server. I understand
> this is a linux issue, not a * issue, except for the following...
> 
> Is the harddisk activity on a standard asterisk install such that I don't
> really have to worry if the power cuts??
> 
> As I understand, if HD activity is minimal, the probability of HD failure is
> significantly reduced.
> 
> P.S. Power regulation is not needed, only protection against instantaneous
> power loss.
> 
> Mike Sander
> 
> --
> No virus found in this outgoing message.
> Checked by AVG Anti-Virus.
> Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 


-- 
Is it something someone said, was it something someone said?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Florz patch for zaphfc

[Matt Riddell]

Nils Segerdahl wrote:
Im running bristuff-0.2.0-rc2b with Florians patch.
4 Billion hfc cards in ptp mode.
Works like a charm.
4 billion hfc cards!  Wow that must be some server :)
Oh a brand name - I guess I missed the capital letter.
hehe
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Can anyone recoment T1/PRI provider in South Ontario?

[Sergey Kuznetsov]

MCI does not provide voice trunks T1/PRI by itself. They resell it as a
add-ons to their IP solutions.
Sprint is expensive. Bell is quite expensive as well. Allstream quite
better in price. ISPTel is the least expensive
one but their customer support is not one of the best.

The best way to find rates for such lines to go to CRTC site and check
the tariffs for that.


All the Best!
Sergey.

Andrew Kohlsmith wrote:

  First things first -- don't reply to a message about something COMPLETELY 
different, erase everything and start your new message.  Just click on the 
"To" and start your new message.  

When you reply and erase everything you are unintentionally placing your 
message in the middle of an existing message thread.  This causes your 
message to get "buried" and far fewer people actually see it.  You don't see 
this because you are using a mail client that has no concept of message 
threads.

http://www.mixdown.ca/~andrew/dump/threaded_email.png is what a mailing list 
looks like to most people, and you can see why replying to a message, erasing 
its contents and starting an entirely new email about a different topic is 
frowned upon (yours is the highlighted message).

Having said that, to your answer:

On January 21, 2005 12:20 am, Robert Augustyn wrote:
  
  
I am looking for a good provider of T1/PRI in Windsor,
Ontario.

  
  
You have many options in large cities.
Bell, Group Telecom(360 networks), AT&T(Allstream), Telus, Sprint, 
MCI(UUnet)...  There may also be a dozen more "little guys" in your area.  
Get a few quotes, I find Bell is actually half-assed competitive when they 
have to be.

Things to consider in your quotes received:
- inbound or two-way call completion
- Number of DIDs per DID/PRI order
- # of #s received for incoming calls (4, 7, or 10 usually)
- If they restrict the PRI signaling in any way
- telephone number "fallback" if the PRI is down (i.e. where do the calls go)
- 911/e911
- capability to set callerID/ANI to any DID you are leasing
- ability to port existing numbers to the PRI as DIDs
- charges for changing anything above once set up

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
  




___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] UPS for Asterisk

[Mike Sander]

I'd considering an UPS backup system for my Asterisk server. I understand
this is a linux issue, not a * issue, except for the following...

Is the harddisk activity on a standard asterisk install such that I don't
really have to worry if the power cuts??

As I understand, if HD activity is minimal, the probability of HD failure is
significantly reduced.

P.S. Power regulation is not needed, only protection against instantaneous
power loss.

Mike Sander

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Florz patch for zaphfc

[Nils Segerdahl]

On Sun, 23 Jan 2005, Stuart Hirst wrote:

> Has anyone had any success using the Florz patch for zaphfc ?
>
> I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
> lines however the users are complaining of crackles on the line which I am
> assuming is related to the IRQ issues raised by Florz.
>
> I have tried to use the patch but it errors trying to patch zaphfc.h
>
> Any help would be appreciated.
Im running bristuff-0.2.0-rc2b with Florians patch.
4 Billion hfc cards in ptp mode.
Works like a charm.
Even spandsp for receiving faxes works.
Pelase describe your problem in more detail.


/Nils

Nils Segerdahl
---
Upsala Systemkonsult, UPSYS AB Telefon:(+46) (0)18 56 80 41
Glunten, 751 83 UppsalaMobil: (+46) (0)703 55 65 03
http://www.upsys.seFax: (+46) (0)18 56 80 49
---
Jan 24  Eskimo Pie patented by Christian Nelson, 1922
Jan 24  Gold discovered in California at Sutter's Mill, 1848
Jan 24  DG Nova introduced, 1969
---

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] VoIP Providers and Backbone Servers

[lonnie]

Hello All,

Well, my explorations in to the world of VoIP is proving fruitful and in
the near future I am hoping to have my small VoIP online service up and
running ready to help promote the industry and hopefully gain a few
customers in the process.

Additionally, I will soon have my IAX and SIP softphone ready that will
handle video, audio, and text communications.

I am looking for quality and fair priced service providers so that I can
add some of thier servies to my VoIP service which will start with an
Asterisk PBX and some reliable Billing software (still trying to decide,
but Trebas or ASTPP looks like it will work for me to get started.)

In particular I will be providing Phone-Phone, Phone-PC, PC-PC, and
PC-Phone connections.

I am looking for services like

1. PSTN Termination Services (Good International Rates)
2. 800 Tollfree  access line services
3. local, national, and international analog access line services in
addition to my Asterisk PBX, if they exist (for my Phone-* services)

and other services that you think are useful.

Any suggestions or comments are appreciated and if you know of a quality
service that I am looking for then now is the time and I invite your
responses to this email.

Thanks to everyone on this for giving me such great help,
Lonnie

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] call return?

[Mike Sander]

For me this worked straight out of the box with [EMAIL PROTECTED] 0.3



Mike Sander
Operations Manager

Suite 4 / 38-48 Waterloo St
Surry Hills N.S.W 2010
Phone:(02) 8307 8877
Fax:(02)93182254
Mobile:0401 010 289
Email: [EMAIL PROTECTED]
Website: www.corporatebankinginternational.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Polk
Sent: Sunday, 23 January 2005 4:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] call return?

Hi:
Can any one point me in the rite direction on this?
I am using asterisk at home for learning purposes. I am trying to get the 
triditional *69 working.
Has there been any success in getting it to announce the number and get it 
to give you the option to call back?

Chris
- Original Message - 
From: "Diego Ventrice" <[EMAIL PROTECTED]>
To: 
Sent: Saturday, January 22, 2005 8:03 AM
Subject: Re: [Asterisk-Users] softswitch dilemma


>
> Thanks for answering Chad,
>
> Actually, I just want to "Switch" traffic between wholesale providers (my
> customers) which actually terminate
> traffic (or not, some of them have just controllers-softswitches like the
> one Im willing to set up)
> collect CDRs and bill them =)
> I have no gateways of my own (of any kind) so Im not originating nor
> terminating calls,
> just "switching" traffic is my goal, all this people use h.323 of course.
>
> Any advice would be appreciated.
>
> Thanks  for your help
> D.
>
>
>> Date: Fri, 21 Jan 2005 22:23:58 -0600 (CST)
>> From: "Chad Whitten" <[EMAIL PROTECTED]>
>> Subject: Re: [Asterisk-Users] softswitch dilemma
>> To: 
>> Message-ID: <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset=iso-8859-1
>>
>> are you looking to do actual pstn to voip termination? if so, then you 
>> are
>> gonna need ss7, cama and imt trunks - things which asterisk doesnt
>> necessarily support.
>>
>> now if you just want to buy pri/t1 from the local telco and sell voip
>> services off an asterisk server that gets back to the pstn over these
>> pri/t1's, then yes, asterisk can do this.
>>
>>
>> Diego Ventrice said:
>> > Hello everybody,
>> >
>> >
>> > Im new to the list and also new to asterisk, Im wondering if I could 
>> > set
>> > up asterisk as a softswitch, I guess for what I've been reading that It
>> > could be possible but almost all the info and documentation Ive found 
>> > so
>> > far is about asterisk as a PBX, etc.
>> >
>> > Im willing to set a small voip wholesale traffic bussiness and Im not
>> > quite sure asterisk is the right chocie for that. An asterisk-ser or an
>> > asterisk-vocal combination may be the answer ?
>> >
>> >
>> > Thanks in advance for any help.
>> > Diego
>>
>>
>> -- 
>> Chad Whitten
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005
 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] IAXy's apparantly failing in the field

[Jean-Michel Hiver]

Michael Giagnocavo wrote:
Yes, the IAXy has faults, but until other IAX2 devices ship, it's the
only game in town. I know that the Farfon device will be out soon and
we'll be able to judge its quality at that time.
   

Or any PA168 phones, which are already out, and support IAX2, SIP, H323,
MGCP and N2P. (I've got one on my desk here as do a few others, and it works
great.) 
 

I want one of them!
Which model is it? Did you have to do any software upgrade? How much 
does it cost?

Cheers,
Jean-Michel.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Autio cut off at beginning of call

[Reid Forrest]

I posted this question a while back, and I'm posting again in hopes that
someone has some ideas. Sorry if you've already seen this.

When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom,
VoicePulse Connect) I often find that after the call is answered the first
few seconds of audio are cut off (i.e. I don't hear the called party). This
usually results in the called party saying "hello Hello???" until I hear
them.

Has anyone else experienced this problem and found a cause or fix? My
internal calls are perfect. It's just Internet-terminated calls that have the
problem. Someone wrote in response to the last post saying that the audio
path probably wasn't set up yet. I think this is the symptom, but I'm
wondering what's the cause, and if there's a fix.

Surely I'm not the only one who's having a huge problem with this. Can anyone
help?

Thank you,
Reid Forrest, CISSP
Max-IS, Inc.
[EMAIL PROTECTED]
ofc: 407.786.9600 x1200   cell: 321.439.8903
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Anybody a patch for oss/alsa to not constantly hog the sound card?

[Bruno Hertz]

The subject says it all. After digging through latency and other issues
with all kinds of linux softphones, I've found that only * works alright
for me as a VoIP client.

Problem now is that, unlike other apps, chan_oss resp. chan_alsa grab
the card once and won't release it until shutdown, while other clients
are friendly enough to grab the card only on calls.

So, before getting lost in a regular coding frenzy, there isn't by
chance any of you who already patched either of those chans to behave a
little more cooperative?

Thanks, Bruno.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Power Alarm Error - Help

[Matt Riddell]

Martin Keding wrote:
Yes, The card is working fine most of the time. It just gets this message on
occasion and then Asterisk shuts down. I debating putting surge suppressors
on the PSTN lines. Could this be caused but a voltage issue from the Telco?
I was told the other day on IRC that telephone line surge protectors 
would only protect against *huge* voltages.

Maybe there is such a thing as a power conditioner for telephone lines?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Any experience with Sangoma cards?

[Robert Augustyn]

Hi,
I am considering A101/102/104 cards for my asterisk installations.
Has any of you used these or any Sangoma cards in such environment?
Any thoughts?
How do they stack up against Digium cards?
Any input would be greatly appreciated.
robert


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Brian Capouch]

Steve Kann wrote:
Actually, the fatal issue is that asterisk's chan_oss or chan_alsa grabs 
the sound device, so iaxclient can't do so.

I can't run it anymore (I used to could. . . ) even on a machine that 
*isn't* running Asterisk.

I haven't changed anything else on my machine, so I think it's somehow 
mis-aiming itself wrt the audio devices on the machine.

At my site things degraded gradually: last summer things worked OK, but 
it only used hoggy codecs.  Once ilbc was added it ran as long as I 
stayed a mile away from that codec.

The latest version doesn't work at all.  When I try to call I get lots 
of errors on the "console:"

ortAudio error at opening separate output stream: Host error.
PaHost_OpenStream: could not open /dev/dsp for O_RDONLY
PaHost_OpenStream: ERROR - result = -1
PortAudio error at opening separate input stream: Host error.
PaHost_OpenStream: could not open /dev/dsp for O_RDONLY
PaHost_OpenStream: ERROR - result = -1
PortAudio error at opening separate input stream: Host error.
PaHost_OpenStream: could not open /dev/dsp for O_RDONLY
PaHost_OpenStream: ERROR - result = -1
PortAudio error at opening separate input stream: Host error.
And the bottom of the interface says, "Can't start audio."
The IAX part appears to be working OK, though, and the little microphone 
VU meter bounces along like it sees *some* kind of audio, but nothing 
comes out the speakers.

FWIW.
B.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Florz patch for zaphfc

[Stuart Hirst]

Has anyone had any success using the Florz patch for zaphfc ?

I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN
lines however the users are complaining of crackles on the line which I am
assuming is related to the IRQ issues raised by Florz.

I have tried to use the patch but it errors trying to patch zaphfc.h

Any help would be appreciated.


Regards,

Stuart
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Steve Kann]

On Jan 23, 2005, at 9:11 AM, Bruno Hertz wrote:
On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:
seem like some kind of port issue...
Actually, the fatal issue is that asterisk's chan_oss or chan_alsa 
grabs the sound device, so iaxclient can't do so.

Probably. Both try to set up listeners on the IAX port
(4569 for IAX2). Disable or reconfigure one of them to
bind to a different port, whichever you want to answer
on it.
In the normal case, asterisk will start first, and get the port, then 
iaxclient will grab a transient port.  This all works out OK, since the 
port that iaxclient uses doesn't matter unless you want to receive 
calls on it without registration.

Also, don't forget to disable chan_alsa and chan_oss in
modules.conf. When running another client you won't want
the * console hogging your soundcard.
Right, that's the thing that will make it work.
-SteveK
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Bandwidth, again, can someone check my math?

[Andrew Kohlsmith]

On January 21, 2005 12:26 pm, Jay Milk wrote:
> G.711, at 64kpbs has a rated network load of 88kbps.
> So for each second of conversation, about 11KB are crossing the wires in
> each direction.

88kbps = 88*1024 bps / 8 bits/byte =11kB/sec, yes, in each direction.

> That means for a minute of two-way conversation, 1.3MB of data are
> transferred?

Yes, if you take the transmit and receive streams separate.  660kB in each 
direction.

> That means for each GB of bandwith, callers can leave almost 800 minutes
> worth of voice-messages?

Seems right to me.  1024*1024 kBytes / 1320kB/min = 794.4 minutes

> Of course, this gets much better if we can get incoming calls on GSM,
> arriving at something like 2,500 minutes/GB.

And even better if you can get VAD support into * so that it isn't sending 
back 660kB of silence per minute.

> Is that correct, or did I mess up a decimal point somewhere?

Seems right.

-A.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Power Alarm Error - Help

[Martin Keding]

Yes, The card is working fine most of the time. It just gets this message on
occasion and then Asterisk shuts down. I debating putting surge suppressors
on the PSTN lines. Could this be caused but a voltage issue from the Telco?

Martin 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, January 23, 2005 10:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Power Alarm Error - Help


> I have been getting the following message in Asterisk and it shuts 
> Asterisk down, needing a reboot.
> 
> "Power alarm on Module 2"
> 
> I have
> (1) TDM400P with (2) FXS & (2) FXO cards
> (1) X100P card
> 
> Any ideas?
Since nobody answered, I'll guess something :)

Did you plug the power on the TDM400P ?  since you have FXS ports, you need
to plug it in ___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Bandwidth, again, can someone check my math?

[Jay Milk]

Bump -- anyone?

> -Original Message-
> From: Jay Milk [mailto:[EMAIL PROTECTED] 
> Sent: Friday, January 21, 2005 11:26 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Bandwidth, again, can someone check my math?
> 
> 
> I want to put a single voice-mail box on a remote server, 
> where I have metered bandwidth.  Before I do this, I want to 
> make sure it's feasible. Could someone confirm the following 
> math for me?
> 
> G.711, at 64kpbs has a rated network load of 88kbps.  
> So for each second of conversation, about 11KB are crossing 
> the wires in each direction.  
> That means for a minute of two-way conversation, 1.3MB of 
> data are transferred? That means for each GB of bandwith, 
> callers can leave almost 800 minutes worth of voice-messages?
> 
> Of course, this gets much better if we can get incoming calls 
> on GSM, arriving at something like 2,500 minutes/GB.
> 
> Is that correct, or did I mess up a decimal point somewhere?

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Eric Wieling]

Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.

That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Actually the issue seems to be more of a "graphics drivers lock 
interrupts for long amounts of time, causing problems with Asterisk". 
 Not specific to X.  A simple VESA Frame Buffer can cause the same 
problems.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Power Alarm Error - Help

[Matt Riddell]

Michael K. Rodriguez User wrote:
I had a similar problem with power.
I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up
and asterisk would load but the T1s on the Quad T1 card failed to come up. I
placed a loop on the card and still no change. Finally, I removed the UPS
and the T1s came up.
Do know if this will help you, but the T1 card seems to be delicate with
power.
Hmmm strange.  Makes you wonder if this is the problem that is occuring 
with the HP machines...

--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dialogic D/4PCI

[Richard Lyman]

Steve Underwood wrote:
Henry Devito wrote:
Is there ant chance of the Dialogic card model D/4PCI working with 
asterisk ?


Word of caution:  Even if you can buy the drivers and make this card 
work with *, it is not meant to plug directly into a CO -48vdc talk 
battery and 90-130vac ring voltage delivered by your phone company.  
These cards were designed to be used behind a pbx.  Most PBX's only 
deliver -24vdc on a analog line now and between 72v and 90vac for 
ring voltage.  Not to say the D/4PCI or D/4PCIU will not work on a 
standard line and not have any problem, just a thought just incase 
you do get this to work and something blows up or gets ruined.  
You'll know why.  Have a great day.

Rubbish. That Dialogic card is designed and approved for use with PSTN 
lines. The only Dialogic cards I know of which are purely PBX oriented 
are their MSI cards, which only *provide* 24V and a lowish ring voltage.

That said, the D/4PCI is useless with *, even if you buy the drivers. 
It is not a full duplex cards. The * drivers are for the JCT cards.

Regards,
Steve
don't forget the older AMX cards.  
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Power Alarm Error - Help

[Michael K. Rodriguez User]

I had a similar problem with power.
I connected Asterisk to a Belkin UPS 1200VA and the the server would boot up
and asterisk would load but the T1s on the Quad T1 card failed to come up. I
placed a loop on the card and still no change. Finally, I removed the UPS
and the T1s came up.

Do know if this will help you, but the T1 card seems to be delicate with
power.

-Michael


On 1/23/05 10:31 AM, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
wrote:

>> I have been getting the following message in Asterisk and it shuts Asterisk
>> down, needing a reboot.
>> 
>> "Power alarm on Module 2"
>> 
>> I have
>> (1) TDM400P with (2) FXS & (2) FXO cards
>> (1) X100P card
>> 
>> Any ideas?
> Since nobody answered, I'll guess something :)
> 
> Did you plug the power on the TDM400P ?  since you have FXS ports, you
> need to plug it in
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20Brings and answers but not hear voice

[Jerry Geis]

Here is the console screen.
Starting simple switch on Zap/1-1
Executing Dial("Zap/1-1", "SIP/403") in new stack
Called 403
SIP/403-9c60 is ringing
SIP/403-9c60 answered Zap/1
Spawn extension (smvoice-incoming, 403, 1) exited nonzero on Zap/1-1
Hangup Zap/1


I have a grandstream 101 that is calling an extension on Zap/1 of a TDM20B.
The grandstream 101 can call another grandstream 101 at a different 
extension- that works fine.
The two phones on TDM 20B can call each other.- no problem.When I call 
the TDM20B Zap/1
from the grandstream phone it rings - I answer and I dont hear any voice.

for the grandstream I have tried allow=all for the codes but made no 
difference.

Any ideas on what I am missing?
Thanks,
Jerry
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: Best VPN server for * and woad warriors using windows?

[Tom Ivar Helbekkmo]

Doug Lytle <[EMAIL PROTECTED]> writes:

> I've had very good success with OpenVPN
> http://sourceforge.net/projects/openvpn

Me too, and I'd certainly use it in the original poster's stead.
However, he specifically said that he must have an IPSEC tool, and
OpenVPN is not IPSEC.

-tih
-- 
Tom Ivar Helbekkmo, Senior System Administrator, EUnet Norway Hosting
www.eunet.no  T +47-22092958 M +47-93013940 F +47-22092901 FWD 484145
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?

[Doug Lytle]

Remco Barende wrote:
What would be the best / easiest VPN software solution. I would like 
to install vpn software on the * server for roadwarriors to connect to 
with laptops running windows. Ideally the vpn solution will not 
require any additional software on the client side but will use IPSEC.

Remco,
I've had very good success with OpenVPN 
http://sourceforge.net/projects/openvpn

Doug
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile

[Peer Oliver Schmidt]

Stephan Schiessling wrote:
Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
As written before, this did the trick for an installation based on the 
bristuff install.sh skript. However, on another system I am running the 
DEB packages provided by http://www.marlow.dk.

That system still does not provide the caller-id to the sip phone.
Any and all help is greatly appreciated.
--
Best regards
Peer Oliver Schmidt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Power Alarm Error - Help

[timebandit001]

> I have been getting the following message in Asterisk and it shuts Asterisk
> down, needing a reboot.
> 
> "Power alarm on Module 2"
> 
> I have
> (1) TDM400P with (2) FXS & (2) FXO cards
> (1) X100P card
> 
> Any ideas?
Since nobody answered, I'll guess something :)

Did you plug the power on the TDM400P ?  since you have FXS ports, you
need to plug it in
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Matt Riddell]

Kenneth Long wrote:
You really do not want to run Asterisk and X-Windows
on the same box.

That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.
are there any other issues, besides loading, to not
run
x-windows at the same time?
Yes.
Whenever you scroll any windows you get clicks on any calls that are in 
progress.  Even on FVWM.

--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Kenneth Long]

> 
> You really do not want to run Asterisk and X-Windows
> on the same box.

That I understand... but this is not a production
machine. Loading is not an issue. I'm using icewm.

are there any other issues, besides loading, to not
run
x-windows at the same time?






__ 
Do you Yahoo!? 
Yahoo! Mail - 250MB free storage. Do more. Manage less. 
http://info.mail.yahoo.com/mail_250
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] grandstream sip phone calling Zap/1 on TDM20B rings and answers but not hear voice

[timebandit001]

Could you give us the output of the console when you try the call ?

That would help us to point you in the right direction.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] How to debug core-file

[hhandresen]

Hi
I'm running safe_asterisk, but get core-files in /tmp - how do I debug 
them ?

I know gdb asterisk core.12370
and bt full
But it didn't show anything usefull for me.
Can anyone help me ?
(Running asterisk 1.0.2 with ast_data
/Hans-Henrik
-
Last from bt full:
priority=200, callerid=0x81b8e90 "Dial", action=1134845864) at 
pbx.c:1384
e = (struct ast_exten *) 0x81bdcf0
sw = (struct ast_switch *) 0x0
data = 0x0
newstack = 1
res = 1134845864
status = 4
incstack = {0x0 , 0x4000 , 0x4025ba20 "", 0x0, 0x0, 0x8fd0ee8 
"1106487988.3352",
  0x43a4775c "1106487988.3352", 0x43a45fa8 "SIP/000b82027e34-0205", 
0x46600d84 "ä\016`Ft\214\a\b¨_¤C", 0x808f25f "\213\233à\003",
  0x8fd0ee8 "1106487988.3352", 0x43a4775c "1106487988.3352", 0x1f 
, 0x4025ba20 "", 0x0, 0x0, 0x0, 0x400382ae 
"\201ÃÞC", 0x0,
  0x4025ba20 "", 0x46600d1c "000b82027e34", 0x0, 0x62303030 , 0x32303238 ,
  0x34336537 , 0x0 , 0x808fcfa "\213\233à\003", 0x8fd0ea8 "´ªóAdö\002", 0x0, 0x0, 0x0}
passdata = "zap/g1/00551138856342|120|rtS(10883)", '\0' 

stacklen = 0
tmp = "\e[1;36;40mDial\e[0;37;40m\000m\000;40m\00040m", '\0' 

tmp2 = "\e[1;35;40mSIP/000b82027e34-0205\e[0;37;40m", '\0' 

tmp3 = 
"\e[1;35;40mzap/g1/00551138856342|120|rtS(10883)\e[0;37;40m\000accountcode:102190|UserID:3456|src:33225075|srcip:217.157.177.77|ConnectPrice:30|PeakPrice:60|RateID:55|CustomerID:30001|DestNameInt:Brazil_São"...
#7  0x08078c74 in ast_pbx_run (c=0x43a45fa8) at pbx.c:1879
digit = 0 '\0'
exten = '\0' 
pos = 0
waittime = 1180700108
res = 0
#8  0x080804e1 in pbx_thread (data=0xfffc) at pbx.c:2102
No locals.
#9  0x40033f60 in pthread_start_thread () from /lib/i686/libpthread.so.0
No symbol table info available.
#10 0x40207327 in clone () from /lib/i686/libc.so.6
No symbol table info available.

Last from bt:
(gdb) bt
#0  0x4019f1f9 in free () from /lib/i686/libc.so.6
#1  0x080568bd in ast_frfree (fr=0x4025afd8) at frame.c:222
#2  0x0805fd54 in ast_channel_bridge (c0=0x43a45fa8, c1=0x8873018, 
config=0x465fc73c, fo=0x465fbe8c, rc=0x465fbe90) at channel.c:2761
#3  0x4139c878 in ast_bridge_call (chan=0x43a45fa8, peer=0x8873018, 
config=0x465fc73c) at res_features.c:342
#4  0x43032239 in dial_exec (chan=0x43a45fa8, data=0x465fc73c) at 
app_dial.c:1003
#5  0x08076b11 in pbx_exec (c=0x43a45fa8, app=0x81b8e90, 
data=0x465fec6c, newstack=148422744) at pbx.c:471
#6  0x0808009d in pbx_extension_helper (c=0x43a45fa8, context=0x43a46100 
"dialout", exten=0x465fec6c "zap/g1/00551138856342|120|rtS(10883)",
priority=200, callerid=0x81b8e90 "Dial", action=1134845864) at 
pbx.c:1384
#7  0x08078c74 in ast_pbx_run (c=0x43a45fa8) at pbx.c:1879
#8  0x080804e1 in pbx_thread (data=0xfffc) at pbx.c:2102
#9  0x40033f60 in pthread_start_thread () from /lib/i686/libpthread.so.0
#10 0x40207327 in clone () from /lib/i686/libc.so.6

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Philipp von Klitzing]

Hi!

> I can run iaxcomm by itself...and I start up Asterisk
> on it own...
> 
> But if I start Asterisk first, then launch iaxcomm
> I get this error:

You really do not want to run Asterisk and X-Windows on the same box.

Cheers, Philipp


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Delay before dialing extension on Zap channel

[Ronan Mullally]

Hi,
After using Asterisk with a SIP hardphone for a couple of weeks I've just 
installed a TDM400P card.

My hardphone - a 7940 - allows me to use a dialplan to decide when a 
particular extension is complete and automatically trigger dialing.  This 
works well with my internal extensions, which are all of the form "Z00".

When trying to dial these extensions from a handset connected to a Zap 
channel there appears to be a delay of about 4 seconds between the time I
dial "Z00" and the time asterisk decides I've finished dialing and 
connects me.

Is there any way to reduce this delay?  I'd ideally like asterisk to dial 
the extension as soon as it matches a valid extension.

-Ronan
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] can iaxcomm run on the same server as Asterisk?

[Bruno Hertz]

On Sat, 2005-01-22 at 23:56 -0800, Kenneth Long wrote:

> seem like some kind of port issue...

Probably. Both try to set up listeners on the IAX port
(4569 for IAX2). Disable or reconfigure one of them to
bind to a different port, whichever you want to answer
on it.

Also, don't forget to disable chan_alsa and chan_oss in
modules.conf. When running another client you won't want
the * console hogging your soundcard.

Regards, Bruno.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?

[Brian Roy]

On Sun, 23 Jan 2005 10:33:14 +0100 (CET), Remco Barende
<[EMAIL PROTECTED]> wrote:

> I would like to
> install vpn software on the * server for roadwarriors to connect to with
> laptops running windows. 

OK, take a hard look at this before you get too far. Installing VPN
software *on* the Asterisk box is not a good idea. Now, you haven't
explained the volume of users on the box, or the availability needs of
the box, but either way, this is bad practice. The term "roadwarriors'
makes me think this is for a business.

> There are numerous vpn server daemons around and I found many messages
> about some of them using tcp/udp etc and instead of trying them all out
> hopefully someone can recommend one?

If you want IPSec, take a look at OpenWall. If you must run this on
your asterisk box, so be it.

Now, if I were you, I would take this opportunity to install a good
Linux based firewall solution that sits in FRONT of the asterisk
server. I can't stress this enough. Take a look at m0n0wall. It has
vpn support (ipsec and pptp) built in, and it will run on nearly
anything. Put this on a machine by itself.

http://m0n0.ch/wall/

> 
> (I guess this would make a useful wiki page too).
> 
> Thanks!! Remco
>

Hope this helps!

-Chuji
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] PIX!!!!!

[Roy Sigurd Karlsbakk]

When we first has the sip fixup enabled, it worked just as you 
described. I think what what happening is as follows:
1. Phones are configured for NAT
2. Cisco PIX "handles NAT" by rewriting headers so the phone doesn't 
appear to be NATted (for SIP proxies that may not support natted 
devices)
3. Asterisk was expecting NAT headers because of nat=yes
FYI: I had to turn off the "NAT handling" in our Pix, since it didn't 
seem to work behind double NAT

roy
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Anyone know where a good source of mailing liststats might be found?

[Jim Van Meggelen]

Thanks to everyone who provided feedback.


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

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.7.2 - Release Date: 21/01/2005
 

___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Outbound analog dialing with Internet Line Jack (fwd)

[Gary]

On Fri, 21 Jan 2005 14:24:39 -0500 (EST), Hayden Myers wrote:

>I've been trying to setup asterisk with an Internet Line Jack card for
>sometime.  I've been successful in configuring asterisk to handle incoming
>calls, make calls between sip phones, call the asterisk demo, and even
>answer the phone with a sip client.  I've been using x-lite as a sip
>client for my testing.  I'm getting pretty close to how I want things to
>work, yet I've still got one major problem.  I can't figure out how to
>place outgoing calls through the analog line the internet line jack is
>connected to.  I'd also like to retrieve caller-id information, but
>haven't been able to do so.  
>
>Hayden Myers   

forget it,
i jst wont work.
.


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile

[Peer Oliver Schmidt]

Hello Stephan,
Another way is to set the callerid in your extensions.conf via exten 
=> 807440,2,SetCIDNum(0${CALLERIDNUM}). 
>
> Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
>
This did the trick. I will go and update the Wiki,,,
Thanks and have a good weekend.
--
Best regards
Peer Oliver Schmidt
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaphfc no callerid incoming to SIP phone but visible in logfile

[Stephan Schiessling]

Try the variable PRI_NETWORK_CID instead of CALLERIDNUM
Peer Oliver Schmidt wrote:
Jens, thanks for the feedback.
>>I've added a ZAPHFC card to my CAPI based system. Calls coming in via
>>ZAPHFC do not forward the caller id to the SIP phones. Calls coming in
>>via CAPI do forward the caller id to the SIP phones.
I think you didn't set usecallerid=yes in your zapata.conf? 

Added it, rebooted, no change. (Before, I just had pritrustusercid = 
yes, only.)

Another way is to set the callerid in your extensions.conf via exten 
=> 807440,2,SetCIDNum(0${CALLERIDNUM}). 

Changed it, now the funny part comes:
extensions.conf
exten => 807440,1,Answer
exten => 807440,2,SetCIDNum(0${CALLERIDNUM})
exten => 807440,3,Dial(SIP/26,20,t)
exten => 807440,3,VoiceMail2(su25)
exten => 807440,103,VoiceMail2(sb25)
exten => 807440,104,Hangup
but the log says:
 -- Accepting call from '1729731418' to '807440' on channel 0/1, span 1
 -- Executing Answer("Zap/1-1", "") in new stack
 -- Executing SetCIDNum("Zap/1-1", "0") in new stack
It does not add the callerid it has two lines above ???
I know there have been some changes to the CID structure sometime 
within Asterisk. But, this is using the bristuff download and install 
script.

The same problem happens using the debian packages (1.0.3) from 
marlow.dk.

Any and all help is greatly appreciated.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Best VPN server for * and woad warriors using windows?

[Brancaleoni Matteo]

Hi,

Il giorno dom, 23-01-2005 alle 10:33 +0100, Remco Barende ha scritto:
> What would be the best / easiest VPN software solution. I would like to 
> install vpn software on the * server for roadwarriors to connect to with 
> laptops running windows. Ideally the vpn solution will not require any 
> additional software on the client side but will use IPSEC.
> (Ofcourse call quality is important)

best if ofcourse some ipsec-based solutions, but that leads
to installing a client on winblow machines.
You can use pptp, ok is not secure as ipsec but is built in
in winblow 98,2k,xp... so on the client you must only
create a new VPN connection (under connections manager)
and you're done.

On the linux side, go to http://poptop.sourceforge.net/dox/
to grab the server.

I think that this is the easiest solutions for a decent
encryption ad ease of use, when using m$ clients.
(hoping you don't need to protect millions $$$ value data :> )

of course ipsec is better, but needs more work to set it
up, on client and on server side.

just my 2 cents,
Matteo


___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Best VPN server for * and woad warriors using windows?

[Remco Barende]

Hi list!
I'm sure the topic has been discussed but I could not find what I was 
looking for.

What would be the best / easiest VPN software solution. I would like to 
install vpn software on the * server for roadwarriors to connect to with 
laptops running windows. Ideally the vpn solution will not require any 
additional software on the client side but will use IPSEC.
(Ofcourse call quality is important)

There are numerous vpn server daemons around and I found many messages 
about some of them using tcp/udp etc and instead of trying them all out 
hopefully someone can recommend one?

(I guess this would make a useful wiki page too).
Thanks!! Remco
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] zaprtc load issue (different that other postings)

[Brian McSpadden]

zaprtc does not work with smp systems, unfortunately. There is some
discussion on the wiki about the bristuff zaprtc module working with
multi cpu systems, however. Link:
http://www.voip-info.org/wiki-Asterisk+timer

Brian


On Sat, 22 Jan 2005 22:37:42 -0800, Spencer Nassar <[EMAIL PROTECTED]> wrote:
> zaprtc 'load' is failing on my machine (the make was fine, same output
> as other posts to this list)
>[EMAIL PROTECTED] zaptelrtc]# make load
>sync
>modprobe zaptel
>insmod ./zaprtc.o
>./zaprtc.o: init_module: Input/output error
>Hint: insmod errors can be caused by incorrect module parameters,
> including invalid IO or IRQ parameters.
>  You may find more information in syslog or the output from dmesg
> 
> I've seen other references to ensuring that rtc is unloaded first, but
> it's not loaded on my machine
>[EMAIL PROTECTED] zaptelrtc]# lsmod
>Module  Size  Used byNot tainted
>zaptel181856   0
>soundcore   7012   0  (autoclean)
>e1000  75744   1
>iptable_nat22744   0  (autoclean) (unused)
>iptable_mangle  2776   0  (autoclean) (unused)
>ipt_REJECT  4632   1  (autoclean)
>ipt_multiport   1176   2  (autoclean)
>ipt_state   1080   3  (autoclean)
>ip_conntrack   29704   2  (autoclean) [iptable_nat ipt_state]
>iptable_filter  2412   1  (autoclean)
>ip_tables  16544   8  [iptable_nat iptable_mangle
> ipt_REJECT ipt_multiport ipt_state iptable_filter]
>microcode   6848   0  (autoclean)
>keybdev 2976   0  (unused)
>mousedev5624   0  (unused)
>hid22276   0  (unused)
>input   6144   0  [keybdev mousedev hid]
>usb-ohci   23176   0  (unused)
>usbcore80928   1  [hid usb-ohci]
>ext3   89960   2
>jbd55060   2  [ext3]
>mptscsih   41780   3
>mptbase43936   3  [mptscsih]
>sd_mod 13360   6
>scsi_mod  112680   2  [mptscsih sd_mod]
> 
> I'm running Redhat ES3 on a dual xeon system
> Kernel is linux-2.4 -> linux-2.4.21-15.EL
> 
> Any ideas?
> 
> Thanks!
> Spencer
> 
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users