date:20040531


On 30/05/2004 at 22:10 Tilghman Lesher wrote:

On Saturday 29 May 2004 16:53, Andy Powell wrote:

If nobody appears to know, it's probable that they haven't done the
experimentation necessary to show one result or another.  If you are
concerned about this behavior, then it falls to you to do the
necessary tests and prove it one way or the other, for the good of the
community.

There's a reason it's called a community -- sometimes you have to
give, instead of just take.

--
Tilghman

Well, there are a number of resposes to this one...

1) If the recommendation (in the bug tracker) is to turn on blocking, but know one 
actually knows what the effect would be, well it's not much of a recommendation is 
it...

2) I think I *have* given something to the community, my getting started guide seems 
to have helped quite a few people get going, I'm in the IRC channel happy to help when 
I can but sometimes *I'd* like some help.

3) I *have* been testing it myself to see the effects.

you assume too much

Andy


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[Asterisk-Users] Updated Zaptel this morning and BOOM CRASH

2004-05-31 Thread Troy Settle


First time around, I just unloaded/reloaded the modules.  The box locked up
tight.  On reboot, I get this:

general protection fault: 
CPU:0
EIP:0010:[c01defb3]Not tainted
EFLAGS: 00010097
eax: f61d4260   ebx: f61d4260   ecx:    edx: f61d425f
esi: f61d4264   edi: f61d4260   ebp: f4de7f14   esp: f4de7ef4
ds: 0018   es: 0018   ss: 0018
Process sh (pid: 387, stackpage=f4de7000)
Stack: 0297 0001 0001 0086 0001 f61d411c 0202
f61d4008 
   0004 f897ebaf 0001  0001 f897eccf f61d411c
0004 
   0008 f61d4008 f6551680   f61d4008 f61d4000
 
Call Trace:[f897ebaf] [f897eccf] [f89c7852] [c01d0349]
[c01d0568]
  [c01d2fa8]

Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 
 0Kernel panic: Aiee, killing interrupt handler!


BTW, this is kernel 2.4.25-gentoo-r2

--
  Troy Settle
  Pulaski Networks
  http://www.psknet.com
  866.477.5638

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RE: [Asterisk-Users] Updated Zaptel this morning and BOOM CRASH

2004-05-31 Thread Troy Settle


 -Original Message-
 From: Troy Settle
 Sent: Monday, May 31, 2004 6:49 AM
 
 First time around, I just unloaded/reloaded the modules.  The 
 box locked up tight.  On reboot, I get this:
 
 general protection fault: 
 CPU:0
 EIP:0010:[c01defb3]Not tainted
 EFLAGS: 00010097
 eax: f61d4260   ebx: f61d4260   ecx:    edx: f61d425f
 esi: f61d4264   edi: f61d4260   ebp: f4de7f14   esp: f4de7ef4
 ds: 0018   es: 0018   ss: 0018
 Process sh (pid: 387, stackpage=f4de7000)
 Stack: 0297 0001 0001 0086 0001 f61d411c 0202
 f61d4008 
0004 f897ebaf 0001  0001 f897eccf f61d411c
 0004 
0008 f61d4008 f6551680   f61d4008 f61d4000
  
 Call Trace:[f897ebaf] [f897eccf] [f89c7852] [c01d0349]
 [c01d0568]
   [c01d2fa8]
 
 Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 
  0Kernel panic: Aiee, killing interrupt handler!
 
 
 BTW, this is kernel 2.4.25-gentoo-r2
 

Oops... Missed one step.  It's not locking up until a few moments (less than
a second?) after running ztcfg.


 --
   Troy Settle
   Pulaski Networks
   http://www.psknet.com
   866.477.5638
 

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Re: [Asterisk-Users] Asterisk Receptionist manager program.

2004-05-31 Thread Peter Corlett

Kyle Hagan [EMAIL PROTECTED] wrote:
[...]
 We are designing it for a touch screen monitor for her to do
 transfers, see whose on the phone and a few other features. Its in
 the development stage and has bugs. but I think its gonna be really
 good.

Warning, heavy use of touch screens causes gorilla arm and careful
ergonomic design of the workspace will be required. Gorilla arm was
discovered in the 1980s and is a reason why touch screens aren't
ubiquitous, but litigation for workplace injuries is both contemporary
and not unusual.

-- 
PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key
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Re: [Asterisk-Users] New Firefly version

2004-05-31 Thread jo

Thanks Adam,
no crash after  installing over 1.5 B3388.  However changing the SIP RTP 
Port is still not accepted.

jo

Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8) with IAX 
 SIP support back in.

Get it from Virbiage site or here's the direct link
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
If it crashes on startup, export your Firefly tree from the registry 
(current user - software - firefly), then delete tree from your 
registry. If that fixes it, send me your exported reg file, there's a 
bug left to do with some wierd reg entry but everyone just deletes it 
instead of sending it to me :|

Transfers will be in the next version - email me any comments, 
requested features, bugs and I'll see what I can do

-Adam
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Re: [Asterisk-Users] New Firefly version

2004-05-31 Thread Steven Thomas


adam - 

can the g729.dll be downloaded somewhere
- is this still required for g.729 support?



Regards,

Steven Thomas








jo [EMAIL PROTECTED]

Sent by: [EMAIL PROTECTED]
31/05/2004 09:19 PM



Please respond to
asterisk-users





To
[EMAIL PROTECTED]


cc



Subject
Re: [Asterisk-Users] New
Firefly version








Thanks Adam,

no crash after installing over 1.5 B3388. However changing
the SIP RTP 
Port is still not accepted.


jo



Adam Hart wrote:

 As Promised, I've released a new version of Firefly (ver 1.8) with
IAX 
  SIP support back in.

 Get it from Virbiage site or here's the direct link
 http://www.virbiage.com/firefly/download/firefly-thirdparty.exe

 If it crashes on startup, export your Firefly tree from the registry

 (current user - software - firefly), then delete tree from
your 
 registry. If that fixes it, send me your exported reg file, there's
a 
 bug left to do with some wierd reg entry but everyone just deletes
it 
 instead of sending it to me :|

 Transfers will be in the next version - email me any comments, 
 requested features, bugs and I'll see what I can do

 -Adam
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Re: [Asterisk-Users] Asterisk mangling faxes

2004-05-31 Thread Steve Underwood

Hi Darren,
Darren Nickerson wrote:
Steve Underwood wrote:
   

Are you sure that approach is free from patent problems? I thought the
ECM protocol was encumbered. If not, that is good news. :-)
   

Steve,
I'm not a lawyer, but I'll give this a shot ;-)
Although particular vendor implementations of ECM may be proprietary, ECM is
a standard outlined in the ITU's Recommendation T.30, so it's no more
encumbered than a whole lot of stuff your rxfax and txfax doodads are
already doing.
 

I don't think that argument holds water. ITU recommendations are filled 
with patented things. I have only implemented FAX features from the 
original FAX standards, which are too old to still have patent problems. 
I am very cautious about adding newer stuff. I have a prototype 
implementation of V.17, but it isn't going into the distribution right 
now, as there appears to be a patent on something about the trellis 
coding (my implementation doesn't do the Viterbi decoding yet, so its 
probably clear of patent issues right now :-) ). T.30 has *many* patent 
claims on it, if you look at the ITU database of patent claims. I think 
one or more relate to ECM.

Regards,
Steve
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Re: [Asterisk-Users] spandsp w/libtiff-3.6.1?

2004-05-31 Thread Steve Underwood

Aaron J. Angel wrote:
Has anyone used spandsp with a patched libtiff 3.6.1 successfully?
 
http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500
Of course not. That is why I keep telling people not to use it. :-)
Regards,
Steve
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[Asterisk-Users] (no subject)

2004-05-31 Thread Girouard, Marc









Wondering if anyone tried to port Asterisk to the Linksys 54G
OpenSource platform?



I am planning to try to port some of the Asterisk code to
that platform and if any once already tried I would like to get in touch with
them . I am thinking on porting the protocol and some other application but not
the Codec itself.



MarcG.

Re: [Asterisk-Users] New Firefly version

get http://www.virbiage.com/firefly/download/g729.zip and follow the 
instructions (you'll need to compile it)

Steven Thomas wrote:
adam -
can the g729.dll be downloaded somewhere - is this still required for 
g.729 support?


Regards,
Steven Thomas


*jo [EMAIL PROTECTED]*
Sent by: [EMAIL PROTECTED]
31/05/2004 09:19 PM
Please respond to
asterisk-users

To
[EMAIL PROTECTED]
cc

Subject
Re: [Asterisk-Users] New Firefly version



Thanks Adam,
no crash after  installing over 1.5 B3388.  However changing the SIP RTP
Port is still not accepted.
jo

Adam Hart wrote:
  As Promised, I've released a new version of Firefly (ver 1.8) with IAX
   SIP support back in.
 
  Get it from Virbiage site or here's the direct link
  http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
 
  If it crashes on startup, export your Firefly tree from the registry
  (current user - software - firefly), then delete tree from your
  registry. If that fixes it, send me your exported reg file, there's a
  bug left to do with some wierd reg entry but everyone just deletes it
  instead of sending it to me :|
 
  Transfers will be in the next version - email me any comments,
  requested features, bugs and I'll see what I can do
 
  -Adam
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Re: [Asterisk-Users] New Firefly version

I'll look at it tomorrow
jo wrote:
Thanks Adam,
no crash after  installing over 1.5 B3388.  However changing the SIP RTP 
Port is still not accepted.

jo

Adam Hart wrote:
As Promised, I've released a new version of Firefly (ver 1.8) with IAX 
 SIP support back in.

Get it from Virbiage site or here's the direct link
http://www.virbiage.com/firefly/download/firefly-thirdparty.exe
If it crashes on startup, export your Firefly tree from the registry 
(current user - software - firefly), then delete tree from your 
registry. If that fixes it, send me your exported reg file, there's a 
bug left to do with some wierd reg entry but everyone just deletes it 
instead of sending it to me :|

Transfers will be in the next version - email me any comments, 
requested features, bugs and I'll see what I can do

-Adam
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[Asterisk-Users] Users in MySQL

2004-05-31 Thread Reto Stauss

I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled (1). During
startup * tells me that it connects to the db, so this should be fine.

Nevertheless I don't see any users from the db when I run sip show users or iax2 
show
users although I configured some.

It is also not possible to call them.

Any hints?

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[Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

We are looking soon at buying a system to deploy asterisk as our 
company's PBX.  We run SuSE here and like it and our asterisk test 
platform is SuSE 9.0 with the 2.4 kernel.

Is anyone running * and the zaptel drivers  under SuSE 9.1?
With the 2.6 kernel?
Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run 
it on an AMD Opteron in 64-bit mode (with whichever kernel is 
acceptable)?

Thank you!
-Michael
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[Asterisk-Users] Meetme + Billing

2004-05-31 Thread Pablo Endres

Hi,

I'm trying to detect and or log the duration a a conference (Meetme). I
need it in order to do some billing for theses services.

Any ideas on how to do it?

I googled around but found nothing.

Thanks in advance

epablo


-- 
Pablo Endres [EMAIL PROTECTED]
ComVoz Communications

USA:   +1 954 343-2085 Ext 199
Venezuela: +58 212 7713195 Ext 199
Colombia:  +57 1 3256840 Ext 199

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RE: [Asterisk-Users] Quicknet PhoneJack Configuration

2004-05-31 Thread asterisk

Hi Kevin,

not sure, but as far as I know, wou're only able to connect a phone to the Quicknet 
Phonejack not a PSTN line!?

Nevertheless, I had the same problem as you are having, but I'm using a Quicknet 
Internet Linejack card.
I have to use the following syntax:

exten = 2501,1,Dial(Phone/phone0,30)

This translated to your envirement would mean:

exten = _9NXXX,1,Dial(Phone/phone0,${EXTEN})

Don't know whether this works at all...

Bjoern


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Chew
Sent: Monday, May 31, 2004 11:52 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Quicknet PhoneJack Configuration


Hi all,
I am still confused about the way to use asterisk with QuickNet Phonejack. If I am not 
wrong, The phonejack card should be using the phone.conf as the asterisk channel. I 
was initially confused with the ZAP channel (The digium card), now that I have found 
out that Phonejack should use the Linux Telephony Devices and its configuration file 
is phone.conf, but the question is I do not know how to configure the extensions.conf 
to call out from sip client to PSTN line. I tried using the 

exten = _9NXXX,1,Dial(Phone/Phone0/${EXTEN})

I keep getting the unable to register channel phone error message. Can anyone please 
paste out a sample extensions.conf file that uses the Quicknet PhoneJack card. Thanks.

 Yahoo! Messenger
- Log on with your mobile phone!


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Re: [Asterisk-Users] spandsp w/libtiff-3.6.1?

2004-05-31 Thread Vlok Stone

I had to go use 3.6.0 before it worked.

On Mon, 2004-05-31 at 03:08, Aaron J. Angel wrote:
 Has anyone used spandsp with a patched libtiff 3.6.1 successfully?
  
 http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500

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Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

2004-05-31 Thread Vlok Stone

I'm running it on a Mandrake 10 w/ 2.6, so it should work. 

On Mon, 2004-05-31 at 13:15, Michael George wrote:
 We are looking soon at buying a system to deploy asterisk as our 
 company's PBX.  We run SuSE here and like it and our asterisk test 
 platform is SuSE 9.0 with the 2.4 kernel.
 
 Is anyone running * and the zaptel drivers  under SuSE 9.1?
 With the 2.6 kernel?
 Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run 
 it on an AMD Opteron in 64-bit mode (with whichever kernel is 
 acceptable)?
 
 Thank you!
 
 -Michael
 
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Re: [Asterisk-Users] Linksys

2004-05-31 Thread Stefan de Konink

Is Asterisk not a *little bit* too much for that processor? SER could be a
better choice?

Stefan

On Mon, 31 May 2004, Girouard, Marc wrote:

 Wondering if anyone tried to port Asterisk to the Linksys 54G OpenSource
 platform?



 I am planning to try to port some of the Asterisk code to that platform and
 if any once already tried I would like to get in touch with them . I am
 thinking on porting the protocol and some other application but not the
 Codec itself.



 MarcG.





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[Asterisk-Users] * on Opteron

2004-05-31 Thread shabanip




anybody has success stories about running * on AMD 
Opteron?

Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

2004-05-31 Thread Julian Pawlowski

Michael George wrote:
Is anyone running * and the zaptel drivers  under SuSE 9.1?
With the 2.6 kernel?
Yes, it runs without any heavy problems here.
Regards,
Julian Pawlowski
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[Asterisk-Users] Re: Users in MySQL

2004-05-31 Thread Stefan Tichy

On Mon, May 31, 2004 at 03:12:44PM +0200, Reto Stauss wrote:
 I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled (1). During
 startup * tells me that it connects to the db, so this should be fine.
 
 Nevertheless I don't see any users from the db when I run sip show users or iax2 
 show
 users although I configured some.

In the sip case it is the consequence of the mode MYSQL_FRIENDS is
implemented. Probably the same with iax2.


 It is also not possible to call them.

When a sip phone registers, the current IP address and other
parameters get updated in the database. This data will be used if
there is a call to this phone.


-- 
Stefan Tichy   [EMAIL PROTECTED]
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[Asterisk-Users] Where is my normal dialtone? With DLINK DG-104S (MGCP)

2004-05-31 Thread Zot O'Connor

I once (for a brief period) had dialtone, but I do know why :)

Otherwise I get a bp-booop-booop sequence.

I cannot tell if this is the D-Link doing this, or asterisk...

Who should be giving solid US dialtone?


My indication.conf says:


[general]
country=us
...
[us]
description = United States / North America
ringcadance = 2000,4000
dial = 350+440
busy = 480+620/500,0/500
ring = 440+480/2000,0/4000
congestion = 480+620/250,0/250
callwaiting = 440/300,0/1
dialrecall =
!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0


-- 
Zot O'Connor [EMAIL PROTECTED]
White Knight Hackers, Inc.

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RE: [Asterisk-Users] Meetme + Billing

2004-05-31 Thread Ray Burkholder

Isn't each call leg represented in the cdr file?  If you set up account
codes properly, it shouldn't be too difficult to script either a
conference duration, or a total call duration to the conference.

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Pablo Endres
 Sent: Monday, May 31, 2004 10:22
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Meetme + Billing
 
 
 Hi,
 
 I'm trying to detect and or log the duration a a conference 
 (Meetme). I
 need it in order to do some billing for theses services.
 
 Any ideas on how to do it?
 
 I googled around but found nothing.
 
 Thanks in advance
 
 epablo
 
 
 -- 
 Pablo Endres [EMAIL PROTECTED]
 ComVoz Communications
 
 USA: +1 954 343-2085 Ext 199
 Venezuela: +58 212 7713195 Ext 199
 Colombia:  +57 1 3256840 Ext 199
 
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 http://www.oneunified.net and is believed to be clean.
 
 


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Re: [Asterisk-Users] Quicknet PhoneJack Configuration files

2004-05-31 Thread Karl Brose

The relevant configuration file is phone.conf and the channel name is 
Phone/phone0
i.e.
exten = 999,1,Dial(Phone/phone0)

Kevin Chew wrote:
Hi all,
I am trying to configure asterisk to work with quicknet phonejack PCI 
card. I tried to serach the internet for the relevant .conf files but 
no results. It seems that for the default configurations is for zaptel 
(I am not sure if it is for digitum card, or can be be used with the 
quicknet cards). So I am appealing for anyone to guide me on how to 
configure the various sip, zapata, zaptel and extensions.conf files to 
be used with the quicknet card. When i tried I keep getting the error 
message No channel type registered for zap then unbale to create 
channel of type zap. everyone is busy at this time. Thank You,

* Yahoo! Messenger 
http://sg.rd.yahoo.com/mail/tagline/?http://sg.messenger.yahoo.com/*
- Log on http://sg.mobile.yahoo.com/sms/msgr20.html with your mobile 
phone!

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Re: [Asterisk-Users] Quicknet PhoneJack Configuration

2004-05-31 Thread Karl Brose

Hi again,
You can't dial out with a PhoneJack. It's an FXS device only.
For dialing out with a Quicknet product you need the LineJack card.
Kevin Chew wrote:
Hi all,
I am still confused about the way to use asterisk with QuickNet 
Phonejack. If I am not wrong, The phonejack card should be using the 
phone.conf as the asterisk channel. I was initially confused with the 
ZAP channel (The digium card), now that I have found out that 
Phonejack should use the Linux Telephony Devices and its configuration 
file is phone.conf, but the question is I do not know how to configure 
the extensions.conf to call out from sip client to PSTN line. I tried 
using the

exten = _9NXXX,1,Dial(Phone/Phone0/${EXTEN})
 
I keep getting the unable to register channel phone error message. Can 
anyone please paste out a sample extensions.conf file that uses the 
Quicknet PhoneJack card. Thanks.

* Yahoo! Messenger 
http://sg.rd.yahoo.com/mail/tagline/?http://sg.messenger.yahoo.com/*
- Log on http://sg.mobile.yahoo.com/sms/msgr20.html with your mobile 
phone!

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Re: [Asterisk-Users] Meetme + Billing

2004-05-31 Thread Fabian Stelzer

In Latest CVS HEAD MeetMe returns the variable MEETMESECS which is
the number of seconds the user was connected to the conference.
But you can also do some scripting of your own and can make it more specific
to you application.
(only billing with multiple users ... and so on)

- Original Message -
From: Pablo Endres [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 31, 2004 3:21 PM
Subject: [Asterisk-Users] Meetme + Billing

 Hi,

 I'm trying to detect and or log the duration a a conference (Meetme). I
 need it in order to do some billing for theses services.

 Any ideas on how to do it?

 I googled around but found nothing.

 Thanks in advance

 epablo

 --
 Pablo Endres [EMAIL PROTECTED]
 ComVoz Communications

 USA:+1 954 343-2085 Ext 199
 Venezuela: +58 212 7713195 Ext 199
 Colombia:  +57 1 3256840 Ext 199

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Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

On May 31, 2004, at 9:58 AM, Julian Pawlowski wrote:
Michael George wrote:
Is anyone running * and the zaptel drivers  under SuSE 9.1?
With the 2.6 kernel?
Yes, it runs without any heavy problems here.
Any light problems?
-Michael
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[Asterisk-Users] Asterisk and MySQL

2004-05-31 Thread Fabio Donaggio




Hi to all!!!

Here's my problem:

-- Executing Dial("SIP/2002-ba7c", 
"SIP/2000|30|tr") in new stackMay 31 16:26:11 NOTICE[262161]: app_dial.c:536 
dial_exec: Unable to create channel of type 'SIP' == Everyone is busy 
at this time -- Executing VoiceMail("SIP/2002-ba7c", 
"b2000") in new stackMay 31 16:26:11 WARNING[262161]: app_voicemail.c:1517 
leave_voicemail: No entry in voicemail config file for 
'2000' -- Executing Hangup("SIP/2002-ba7c", "") in new 
stack == Spawn extension (from-sip, 2000, 103) exited non-zero on 
'SIP/2002-ba7c'
I followinstructions that I found 
in

http://www.voip-info.org/wiki-Asterisk+voicemail+database

but voicemail not work with my MySql 
database

I'm in your hands

Thanks

[Asterisk-Users] D-Channel Problems

2004-05-31 Thread Jeff Sczpel

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Good day eveyone,

I'm hoping that someone can help me.  Perhapps i'm overlooking the obvious,
 in truth, I hope that I am.  I've scoured the mailing list and google,
 and haven't come up with much.

I have a Digium T400P thats been connected to a channel bank for testing
for some time now.  all has been well.

I've now just had installed a PRI from Allegiance Telecom.  All 23 chanels
are active, and they've told me that the 24th is the dchannel.  switchtype
is Lucent 5ess.  Trunk is esf  b8zs

[/etc/zaptel.conf]
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us

[/etc/asterisk/zapata.conf]
[channels]
switchtype=5ess
context=incoming-pri
signalling=pri_cpe
echotraining=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
echocancel=yes
echocancelwhenbridged=no
group=5
immediate=no
callerid=asreceived
musiconhold=default
channel = 1-23

When I start Asterisk, I get this;
- 
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 10
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 11
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 12
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 13
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 14
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 15
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 16
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 17
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 18
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 19
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 20
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 21
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 22
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 23
May 30 21:54:07 WARNING[1175660480]: chan_zap.c:6174 zt_pri_error: PRI:
Read on 130 failed: Unknown error 500
May 30 21:54:07 NOTICE[1175660480]: chan_zap.c:6905 pri_dchannel: PRI
got event: 5 on span 1
- --

and Asterisk never brings up the d-channel.  No calls are successful
in or out. (cannot create channel of type zap)


ANY ideas?
MUCH Appreciated if anyone can be of assistance.

Thanks,
Jeff
-BEGIN PGP SIGNATURE-
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Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

On May 31, 2004, at 9:15 AM, Michael George wrote:
We are looking soon at buying a system to deploy asterisk as our 
company's PBX.  We run SuSE here and like it and our asterisk test 
platform is SuSE 9.0 with the 2.4 kernel.

Is anyone running * and the zaptel drivers  under SuSE 9.1?
With the 2.6 kernel?
Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run 
it on an AMD Opteron in 64-bit mode (with whichever kernel is 
acceptable)?
It sounds like the 2.6 kernel is not a problem.
Anyone able to verify results on a 64-bit processor?
Also, we might temporarily try running it on Yellow Dog Linux (based on 
RHL9.0) on a PPC (Mac).  I don't know of any reason why this wouldn't 
work but success stories always help... :)

Thanks!
-Michael
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Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

Sorry, I spoke before I read.
The archives showed me that the Opteron and Yellow Dog do not seem to 
be a problem...

Thanks to all who responded!
On May 31, 2004, at 9:15 AM, Michael George wrote:
We are looking soon at buying a system to deploy asterisk as our 
company's PBX.  We run SuSE here and like it and our asterisk test 
platform is SuSE 9.0 with the 2.4 kernel.

Is anyone running * and the zaptel drivers  under SuSE 9.1?
With the 2.6 kernel?
Is * 64-bit safe (i.e. no 32bit assumptions in the code) so I can run 
it on an AMD Opteron in 64-bit mode (with whichever kernel is 
acceptable)?

Thank you!
-Michael
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-Michael
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Re: [Asterisk-Users] Unblocking incoming SIP


On 30/05/2004 at 21:35 Thor Atle Rustad wrote:

I have just set up my first Asterisk, and I have the basics up an
running.
I have set it up with two SIP phones only. I can call between them, and I
can call out to FWD phones. However, on receiving calls from FWD, my
Asterisk blocks the calls with the following message:

May 30 20:19:24 NOTICE[180236]: chan_sip.c:6397 handle_request: Failed to
authenticate user user sip:[EMAIL PROTECTED]. Obviously, I want
FWD users to be able to call me without my registering them first.

Any suggestions would be appreciated.

Thor

Thor,

this is because some oh so clever person decided that the default 'security' option for
sip should be to reject anything that's not in sip.conf

put :

insecure=very

in your fwd definition in sip.conf

It was basically that everything had to authenticate... which the fwd number couldn;t 
because
it wasn;t defined in sip.conf.

Anything that's added to * that breaks how protocols work should be by default OFF not 
ON,
but that's just IMO...

Of course I find it quite funny that it's insecure=very, perhaps it should be:

make-SIP-work-how-it-is-supposed-to=yes

;)

Blah blah moo!

Andy


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Re: [Asterisk-Users] Sipura-spa2000

2004-05-31 Thread Simon Chappell

I have fiddled about and managed to get some of the phones to work, only
the  fixed ones though none of the dect ones will work..
Is there anyway to get the recall button or some other button to work
instead of hook flash ?

Simon



 I had the same problem with a Siemens dect once ( and with Sipura ).
 The problem was solved by adding flash hook time. This is a configurable
 parameter in many dect phones. I added several hundreds of ms and the
 button
 started to work  ( or actually - Sipura was able to 'see' the action ).

 -- Pertti


 Simon Chappell wrote:

 thanks for the reply, i thought it may be a stupid question but if i
 hit either hook buttons i do not get  any result when in a call. if i
 press the hangup button it hangs up, press the pick up button and
 nothing happens :-(

 that is why i thought I was doing something silly or not understanding
 something.

 It is a panasonic dect phone

 Simon

 Richard Neese wrote:

 the off hook / hangup switch should act as a flash button also...
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---
Kind Regards

Simon Chappell
Email : [EMAIL PROTECTED]
WWW   : www.isnsuk.com
Phone : 01403268474
Mobile: 07811409125
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RE: [Asterisk-Users] * on Opteron

2004-05-31 Thread usedcanon




I have used with 
Athlon 64, but noth opteron. Can imagine it being much different though. 


Umar

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of 
  shabanipSent: 31 May 2004 14:55To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] * on 
  Opteron
  anybody has success stories about running * on 
  AMD Opteron?

[Asterisk-Users] Chan Capi Audio Quality Issue...

2004-05-31 Thread Stefano Finetti

Hello all,

I've just finished to install chan_capi with 3 AVM Fritz PCI cards.

It correctly loads the 3 drivers, and * starts without errors.

immediately after * start, audio quality is really fine, but, after the
first incoming call, all incoming audio is broken, trembling and stuttering.

From the other side, audio is still fine.

Basically if i receive a call, who called me hears a fine audio while i hear
only stuttering noise.

After the first call ends, the problem persist on each call until i restart
*.

If no call are received, all goes well with fine audio (i.e. without
incoming calls answered, outgoing calls go just fine as expected.

This is my capi.conf file (don't know if it's important in this case, but
better put than not :-) )

;
; CAPI config
;
;
[general]
nationalprefix=0
internationalprefix=00
rxgain=1.0
txgain=1.0

[interfaces]

mode=immediate
msn=0
incomingmsn=*
;isdnmode=ptmp
controller=1,2,3
softdtmf=1
overlap=1
context=default
echosquelch=1
echocancel=yes
echotail=64
;deflect=12345678
callgroup=1
devices=2,2,2


Any suggestion? i've done a quick search on ml archives but didn't find a
similar problem...

Thanks in advance

--
Stefano

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Re: [Asterisk-Users] Quicknet PhoneJack Configuration

2004-05-31 Thread Eric Wieling

On Mon, 2004-05-31 at 09:33, Karl Brose wrote:
 Hi again,
 You can't dial out with a PhoneJack. It's an FXS device only.
 For dialing out with a Quicknet product you need the LineJack card.

Check the mailing list archives for the limitations in dialing/receiving
calls on the LineJack

To search the Asterisk mailing list archive go to www.google.com and put
site:lists.digium.com in addition to your other query terms.


-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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RE: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Kevin Walsh


 I'm hoping that someone can help me.  Perhapps i'm overlooking the
  obvious, in truth, I hope that I am.  I've scoured the mailing list and
  google, and haven't come up with much.
 
 I have a Digium T400P thats been connected to a channel bank for testing
 for some time now.  all has been well.
 
 I've now just had installed a PRI from Allegiance Telecom.  All 23 chanels
 are active, and they've told me that the 24th is the dchannel.  switchtype
 is Lucent 5ess.  Trunk is esf  b8zs
 
 [/etc/zaptel.conf]
 span=1,1,1,esf,b8zs
 bchan=1-23
 dchan=24
 loadzone=us
 defaultzone=us
 
Note that I don't have any of the equipment you mentioned, so I may
have little or no idea of what I'm talking about.  I thought I'd post
this anyway, as it might fall into the overlooking the obvious
category.

Have you tried different values in the span directive, in your
zaptel.conf?  You may want to try this setting:

span=1,0,0,esf,b8zs

That's all I can think of.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Eric Wieling

On Mon, 2004-05-31 at 09:44, Jeff Sczpel wrote:

 [/etc/zaptel.conf]
 span=1,1,1,esf,b8zs

span=1,0,1,esf,b8zs

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Duane

Andy Powell wrote:
Anything that's added to * that breaks how protocols work should be by default OFF not ON, 
but that's just IMO...
I agree 100%, this has been very frustrating trying to work out why 
Asterisk suddenly stopped accepting calls from FWD and other PSTN based 
services.

I very much preferred the old method, if I didn't want to accept a SIP 
call you just don't have a matching context.

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
In the confrontation between the stream and the rock, the
stream always wins; not through strength, but through persistence.
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Re: [Asterisk-Users] Asterisk and Zaptel for 2.6 kernel

2004-05-31 Thread Julian Pawlowski

Michael George wrote:
Any light problems?
Not even those. A few come from the development version of asterisk but 
not from the operating system environment itself.

Regards
Julian Pawlowski
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RE: [Asterisk-Users] Sipura-spa2000

2004-05-31 Thread Kevin Walsh

Simon Chappell [EMAIL PROTECTED] wrote:
 I have fiddled about and managed to get some of the phones to work, only
 the  fixed ones though none of the dect ones will work..
 Is there anyway to get the recall button or some other button to work
 instead of hook flash ? 
 
Try the following in the SPA-2000's Regional page:

Hook Flash Timer Min: 0.05
Hook Flash Timer Max: 0.5

It works for me.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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[Asterisk-Users] Disclaimer fax number?

2004-05-31 Thread Tony Mountifield

Which is the correct fax number for disclaimers?

http://bugs.digium.com/main_page.php says +1-256-864-0464

http://www.digium.com/bugtracker.html says +1-256-971-6890

Or are they both equally good?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Eric Wieling

On Mon, 2004-05-31 at 10:16, Duane wrote:
 Andy Powell wrote:
 
  Anything that's added to * that breaks how protocols work should be by default OFF 
  not ON, 
  but that's just IMO...
 
 I agree 100%, this has been very frustrating trying to work out why 
 Asterisk suddenly stopped accepting calls from FWD and other PSTN based 
 services.
 
 I very much preferred the old method, if I didn't want to accept a SIP 
 call you just don't have a matching context.

The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].


-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Steven Critchfield

On Mon, 2004-05-31 at 10:16, Eric Wieling wrote:
 On Mon, 2004-05-31 at 09:44, Jeff Sczpel wrote:
 
  [/etc/zaptel.conf]
  span=1,1,1,esf,b8zs
 
 span=1,0,1,esf,b8zs

I'm writing this because you seem to be getting guesses from people who
aren't telling you it is a guess. 

from /etc/zapata.conf
span=span num,timing,line build out (LBO),framing,coding[,yellow]

so you need
span=1,1,0,esf,b8zs

You must take your timing from the line as primary. You don't need to
modify the build out as it isn't going to be more than 133ft from the
smartjack. 

Note that you may have to power cycle the machine to set the timing
properly. Verify timing using zttool. 
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Jeff Sczpel

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Unfortunately, I've tried many different timing and build-out settings.
 Doesn't seem to help a thing.

~jeff

On Mon, 31 May 2004 08:16:37 -0700 Eric Wieling [EMAIL PROTECTED] wrote:
On Mon, 2004-05-31 at 09:44, Jeff Sczpel wrote:

 [/etc/zaptel.conf]
 span=1,1,1,esf,b8zs

span=1,0,1,esf,b8zs
-BEGIN PGP SIGNATURE-
Note: This signature can be verified at https://www.hushtools.com/verify
Version: Hush 2.4

wkYEARECAAYFAkC7U+EACgkQjPSWd97xE06TmwCfWC5RmpxzCqqbaHfvCJ2x/xTMMlsA
n0mVNBVT3E+07m4ZeEwrXo9i4RSR
=5p9y
-END PGP SIGNATURE-




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[Asterisk-Users] Crc4 issues

2004-05-31 Thread Paulo Mannheimer

Hi All,

This is our 2nd E1 client that we try to use crc4 either with the e100p
or with the e405p without luck. 

After some trials, we ask the telco to switch off crc4 on their side and
everything works flawlessly.

Is there anything in the crc4 calculation that may be broken? We took a
look at wct1xxx.c and wct4xx.c but there doesn't seem to be much there
to be fixed (apparently the crc4 calculation is done within the chip
itself).

We also took a look at
http://lists.digium.com/pipermail/asterisk-cvs/2003-September/000126.htm
l but couldn't figure out what bits should we try to set to test other
card options.

Is there any documentation on the card that could help us?

Our zaptel looks like ...

span=1,0,0,ccs,hdb3,crc4

bchan=1-15,17-31
dchan=16

We already tried ...

span=1,1,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4,yellow
span=1,0,0,ccs,hdb3,crc4,yellow


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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Duane

Eric Wieling wrote:
The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
I wonder where they picked that up from, default config perhaps?
--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
In the confrontation between the stream and the rock, the
stream always wins; not through strength, but through persistence.
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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Andres


The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
 

Thats the way we prefer it (the old way).  Its nice to be able to 
publish a sip phone number to anybody out there(for example I can just 
say that my number is sip:[EMAIL PROTECTED]).  When the 
call comes into Asterisk (from whatever SIP source), the [general] 
section tells it to take the call to the Autoattendant in whatever 
context you have defined.  Otherwise we have now lost that possibility.

--
Andres
Network Admin
http://www.telesip.net

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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Andrew Kohlsmith

On Monday 31 May 2004 11:47, Steven Critchfield wrote:
 Note that you may have to power cycle the machine to set the timing
 properly. Verify timing using zttool.

Offhand, why is that?  Will a module unload/load not be sufficient?  I had not 
heard of this until now.

-A.
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Re: [Asterisk-Users] Unblocking incoming SIP

On 31/05/2004 at 10:47 Eric Wieling wrote:

On Mon, 2004-05-31 at 10:16, Duane wrote:
 Andy Powell wrote:

  Anything that's added to * that breaks how protocols work should be by
default OFF not ON,
  but that's just IMO...

 I agree 100%, this has been very frustrating trying to work out why
 Asterisk suddenly stopped accepting calls from FWD and other PSTN based
 services.

 I very much preferred the old method, if I didn't want to accept a SIP
 call you just don't have a matching context.

The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].


which is why everywhere you look in the guides etc people say put something like:

context=boguscalls

in the general section, which (providing you weren't stupid enough to create a
[boguscalls] section worked well... in fact I'll go as far as quoting my own guide:

An important point here, if you do not have a sip aware firewall and are just using 
port forwarding then ensure that your context points to somewhere like invalidcalls. 
If you do not do this then someone could call one of your extensions direct from the 
Internet. If you had an FXO card in the machine, this could lead to them being able to 
make PSTN calls

Those people that didn't realize were more than likely using a guide to set up...

I still stand by the fact that this feature should have been OFF in the first place.

Andy


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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Steven Critchfield

On Mon, 2004-05-31 at 11:14, Andrew Kohlsmith wrote:
 On Monday 31 May 2004 11:47, Steven Critchfield wrote:
  Note that you may have to power cycle the machine to set the timing
  properly. Verify timing using zttool.
 
 Offhand, why is that?  Will a module unload/load not be sufficient?  I had not 
 heard of this until now.

Don't know why, just know that unless the power is cycled on at least 1
of our cards, the timing report in zttool doesn't change. Just an
observation, should make life simple and save your hair.
-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] Unblocking incoming SIP



*** REPLY SEPARATOR  ***

On 31/05/2004 at 11:13 Andres wrote:


Thats the way we prefer it (the old way).  Its nice to be able to 
publish a sip phone number to anybody out there(for example I can just 
say that my number is sip:[EMAIL PROTECTED]).  When the 
call comes into Asterisk (from whatever SIP source), the [general] 
section tells it to take the call to the Autoattendant in whatever 
context you have defined.  Otherwise we have now lost that possibility.



which is another good point :D


Andy


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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Brian Cuthie

Eric Wieling wrote:
On Mon, 2004-05-31 at 10:16, Duane wrote:
 

Andy Powell wrote:
   

Anything that's added to * that breaks how protocols work should be by default OFF not ON, 
but that's just IMO...
 

I agree 100%, this has been very frustrating trying to work out why 
Asterisk suddenly stopped accepting calls from FWD and other PSTN based 
services.

I very much preferred the old method, if I didn't want to accept a SIP 
call you just don't have a matching context.
   

The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].
 

Which is because configs/sip.conf.sample has context=default. So let's 
not blame it on the too many people problem.

I agree that new features shouldn't break old configs.
-brian
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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Brian Cuthie

Andy Powell wrote:
On 31/05/2004 at 10:47 Eric Wieling wrote:

On Mon, 2004-05-31 at 10:16, Duane wrote:

Andy Powell wrote:

Anything that's added to * that breaks how protocols work should be by

default OFF not ON,

but that's just IMO...

I agree 100%, this has been very frustrating trying to work out why
Asterisk suddenly stopped accepting calls from FWD and other PSTN based
services.

I very much preferred the old method, if I didn't want to accept a SIP
call you just don't have a matching context.

The problem is that too many people had a context= in [general] and
didn't realize that incoming SIP calls that didn't match anything would
be accepted and sent to the context= that was specified in [genera].

which is why everywhere you look in the guides etc people say put something like:
context=boguscalls
in the general section, which (providing you weren't stupid enough to create a
[boguscalls] section worked well... in fact I'll go as far as quoting my own guide:
An important point here, if you do not have a sip aware firewall and are just using port forwarding then
ensure that your context points to somewhere like invalidcalls. If you do not do this then
someone could call one of your extensions direct from the Internet. If you had an FXO card in the machine,
this could lead to them being able to make PSTN calls
Those people that didn't realize were more than likely using a guide to set up...

I still stand by the fact that this feature should have been OFF in the first place.
Andy

Except that I *want* anyone to be able to call me directly from the
Internet. That's the whole point -- we're trying to remove the necessity
for a phone-company-like entity in the middle.

Instead, I suggest setting the default context for sip to something like
sip-incoming-default and then include in the dialplan those things you
wish people to be able to call directly.

-brian
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[Asterisk-Users] I want to purchase atleast one used quicknet card

2004-05-31 Thread Scott Edwards

If you have any quicknet cards you are not using, I may be interested in 
them.  I'll discuss terms after I know what you have.

For sake of high shipping costs, I'm not interested in overseas shipments to 
the United States (where I live).  My best resources will be for those that 
have PCI PhoneJacks, or PCMCIA CardJacks.

Please reply to my email address, and not the mailing list.
Thank you.
_
FREE pop-up blocking with the new MSN Toolbar  get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Asterisk-Users] I want to purchase atleast one used quicknet card

2004-05-31 Thread Scott Edwards

If you have any quicknet cards you are not using, I may be interested in 
them.  I'll discuss terms after I know what you have.

For sake of high shipping costs, I'm not interested in overseas shipments to 
the United States (where I live).  My best resources will be for those that 
have PCI PhoneJacks, or PCMCIA CardJacks.

Please reply to my email address, and not the mailing list.
Thank you.
_
FREE pop-up blocking with the new MSN Toolbar  get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

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[Asterisk-Users] line config

2004-05-31 Thread Sean McKay


 I was wondering, might it be possible to setup the following scenario:

SIP/iconnecthere - internal extension 1
SIP/bt - internal extension 2
..

 Have it be transparent. internalext 1 would appear as my iconnect
phone number, as well as internalext 2 would appear as my bt number
so if I pick up extension2 it will only dial out to my bt line,
as with ext 1 would only dial out to iconnecthere. If someone rings
my bt line, ring extentsion 2, then voicemail(user 1000 i.e.), if
someone rings my iconnecthere line ring extension 1, then voicemail
(same voicemail user).

Now to add onto that. Would it be possible to hook both of those
2 extensions into 2 or more ip phones and have the ability to
put a call on hold on one phone while be able to pick it up
(same extension) on other phone. Have the ability to tell me
on one phone that another phone has a specified extension engaged
and also be able to pick up and join in a current call without
call parking, or anything else like that.

thanks
sean
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RE: [Asterisk-Users] Crc4 issues

2004-05-31 Thread Storer, Darren

Hi Paulo,

PM This is our 2nd E1 client that we try to use crc4 either with the
PM e100p or with the e405p without luck.

PM After some trials, we ask the telco to switch off crc4 on their side
PM and everything works flawlessly.

[span=1,1,0,ccs,hdb3,crc4,yellow] looks good as it uses CRC4 and sets the
timing to be synchronised with the clock coming in from your Telco's switch.
You do not mention what sort of switch you are trying to connect to and what
sort of physical cabling (including length) is used for connection to the
Telco (Coax, baluns, 120 Ohm RJ45 etc.)???

On the occasions where CRC4 has proved to be a major problem from Asterisk
to the Telco's switch, bad cable termination on the frame proved to be the
problem and as soon as the connections were re-made properly CRC4 worked
perfectly.

I would also refer you to a recent comment from Critch who advised that
Asterisk systems should be power cycled when changing CRC4 and timing
settings for PRI. I agree with Critch _completely_; you must 'init 6' the
system when you make PRI changes otherwise you will obtain false results and
waste a lot of time.

If the comments above do not help perhaps you could provide a bit more
background information and then someone on the list will be able to assist.

HTH

Darren
--
Comgate
TelcoInternetBroadcast
+44(0)700 COMGATE

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paulo
Mannheimer
Sent: 31 May 2004 17:08
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Crc4 issues


Hi All,

This is our 2nd E1 client that we try to use crc4 either with the e100p
or with the e405p without luck.

After some trials, we ask the telco to switch off crc4 on their side and
everything works flawlessly.

Is there anything in the crc4 calculation that may be broken? We took a
look at wct1xxx.c and wct4xx.c but there doesn't seem to be much there
to be fixed (apparently the crc4 calculation is done within the chip
itself).

We also took a look at
http://lists.digium.com/pipermail/asterisk-cvs/2003-September/000126.htm
l but couldn't figure out what bits should we try to set to test other
card options.

Is there any documentation on the card that could help us?

Our zaptel looks like ...

span=1,0,0,ccs,hdb3,crc4

bchan=1-15,17-31
dchan=16

We already tried ...

span=1,1,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4,yellow
span=1,0,0,ccs,hdb3,crc4,yellow


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Re: [Asterisk-Users] I want to purchase atleast one used quicknet card

2004-05-31 Thread Michael Lingwall

Hey,

Just pay digium the extra couple dollars.

It isnt worth the fight with the quicknet cards and the digiums are by far
more supported easily.

Michael
- Original Message - 
From: Scott Edwards [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 31, 2004 10:38 AM
Subject: [Asterisk-Users] I want to purchase atleast one used quicknet card


 If you have any quicknet cards you are not using, I may be interested in
 them.  I'll discuss terms after I know what you have.

 For sake of high shipping costs, I'm not interested in overseas shipments
to
 the United States (where I live).  My best resources will be for those
that
 have PCI PhoneJacks, or PCMCIA CardJacks.

 Please reply to my email address, and not the mailing list.

 Thank you.

 _
 FREE pop-up blocking with the new MSN Toolbar - get it now!
 http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

 ___
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Re: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Kwijibo

I would start with the basics.
Are you sure the T1 is even operating correctly?
Any errors/slipped seconds, etc?
Maybe they haven't configured their end correctly yet?
It almost sounds as if they have their end looped.  I don't
think Zaptel hardware can detect these.
Just some ideas
Steve
Jeff Sczpel wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Unfortunately, I've tried many different timing and build-out settings.
Doesn't seem to help a thing.
~jeff
On Mon, 31 May 2004 08:16:37 -0700 Eric Wieling [EMAIL PROTECTED] wrote:
 

On Mon, 2004-05-31 at 09:44, Jeff Sczpel wrote:
   

[/etc/zaptel.conf]
span=1,1,1,esf,b8zs
 

span=1,0,1,esf,b8zs
   

-BEGIN PGP SIGNATURE-
Note: This signature can be verified at https://www.hushtools.com/verify
Version: Hush 2.4
wkYEARECAAYFAkC7U+EACgkQjPSWd97xE06TmwCfWC5RmpxzCqqbaHfvCJ2x/xTMMlsA
n0mVNBVT3E+07m4ZeEwrXo9i4RSR
=5p9y
-END PGP SIGNATURE-

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RE: [Asterisk-Users] Galaxy Voice

2004-05-31 Thread Dr. Rich Murphey

If it fails to register, check the sip debug output for:

REGISTER sip:216.229.127.40 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060

If you see 0.0.0.0 in the 'Via' line, try using

nat=yes
externip=your external address

in your *global* section at the head of sip.conf.

I've searched but haven't been able to find where the value is being set to
0.0.0.0.

Cheers,
Rich

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kevin
 Sent: Saturday, May 29, 2004 1:24 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Galaxy Voice
 
 Yes, I did a search and have what I think is the correct configuration.
 I did a google search and I didn't see much.  I was successful in
 getting it to work both inbound and outbound with the exception of the
 notices and warnings.
 
 The config I am using is:
 
 [galaxyvoice]
 nat=yes
 port=5060
 fromuser=12345678
 fromdomain=216.229.127.40
 username=12345678
 type=friend
 secret=12345678
 auth=md5
 host=216.229.127.40
 ;defaultip=216.229.127.40
 reinvite=no
 canreinvite=no
 dtmfmode=rfc2833
 context=inbound-galaxy
 qualify=yes
 disallow=all
 allow=gsm
 allow=ulaw
 callerid=12345678
 incominglimit=2
 defaultexpirey=60
 
 
 -Original Message-
 From: brian k. west [mailto:[EMAIL PROTECTED]
 Sent: Saturday, May 29, 2004 2:58 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Galaxy Voice
 
 Also I think someone posted a galaxy voice config example on the mailing
 list a few weeks back.. have you searched google yet?
 
 bkw
 - Original Message -
 From: Kevin  [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, May 29, 2004 11:04 AM
 Subject: RE: [Asterisk-Users] Galaxy Voice
 
 
  I deeply apologize for the incorrect statement, thanks for taking the
  time to point out the error...your help is appreciated.
 
  -Original Message-
  From: brian k. west [mailto:[EMAIL PROTECTED]
  Sent: Saturday, May 29, 2004 1:31 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Galaxy Voice
 
  First off they are not ERRORS  they are NOTICE and WARNING.
 
  bkw
 
  - Original Message -
  From: Kevin  [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Saturday, May 29, 2004 10:26 AM
  Subject: [Asterisk-Users] Galaxy Voice
 
 
   Has anyone successfully used Galaxy Voice with Asterisk?
  
   I am getting the following SIP errors repeated whether it is or
 isn't
   behind NAT.
  
   May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt:
  Maximum
   retries exceeded on call [EMAIL PROTECTED]
  for
   seqno 104 (Critical Request)
   May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout:
   Registration for '[EMAIL PROTECTED]' timed out, trying again
   May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt:
  Maximum
   retries exceeded on call [EMAIL PROTECTED]
  for
   seqno 111 (Critical Request)
   May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597 sip_reg_timeout:
   Registration for '[EMAIL PROTECTED]' timed out, trying again
   asterisk2*CLI
  
  
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Re: [Asterisk-Users] Linksys

2004-05-31 Thread Tony Hoyle

Stefan de Konink wrote:
Is Asterisk not a *little bit* too much for that processor? SER could be a
better choice?
The asterisk binary alone is larger than the total flash ram space on the linksys.
I really doubt it's going to work
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle [EMAIL PROTECTED]  Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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Re: [Asterisk-Users] Linksys

2004-05-31 Thread Greg Boehnlein

On Mon, 31 May 2004, Tony Hoyle wrote:

 Stefan de Konink wrote:
  Is Asterisk not a *little bit* too much for that processor? SER could be a
  better choice?
 
 The asterisk binary alone is larger than the total flash ram space on the linksys.
 
 I really doubt it's going to work

That assumes that compiling for the MIPS w/ uClibc is going to result in a 
binary that is similar in size to that of an x86 binary, which isn't 
neccessarily going to be true.

There are several other tricks that could be used to reduce the size of 
the binary and associated modules. Just comes down to taking the time to 
do it.

I just picked up a WRT54G yesterday to replace a failing Gateway Router 
and found myself asking the exact same question. I.E. how can I use the 
WRT54G to fit in with my Asterisk system?

SER might be an interesting option, though! :)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] * on Opteron

2004-05-31 Thread Greg Boehnlein

On Mon, 31 May 2004, usedcanon wrote:

 I have used with Athlon 64, but noth opteron. Can imagine it being much
 different though.

I'll let you know in a couple of weeks when my Dual Opteron workstation is 
finished.


-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Snom and multiple lines

2004-05-31 Thread Dennis Engdahl

Sorry that I didn't do more research first.

It seems that the problem here is discussed on Mantis as bug report 652.
Mark states that * now supports subscribe/notify, and therefore the problem
is resolved.  However, the moment I configure a Snom 200 for more than one
line, all the configured buttons light up and stay on. rather than flashing
for calls, etc.

I understand on the 652 discussion, that there is a problem identifying
whether or not an extension or a piece of equipment is busy to control
these lights (since a single extension number can ring multiple pieces of
equipment, etc.), the real question for a secretay/boss situation is whether
ANY piece of equipment assigned to the extension is busy or not.  So, first,
why do the lights stay on, and secondly, can they light when anyone is using
that extension?

Thanks.  This is really a problem for us.

Not only do we need the secretay/boss key system arrangement, but a
traveling technician would like to be able to add his SIP extension to
someone else's phone when he is working at their station.

Any help would be appreciated.

Thanks,

Dennis Engdahl
[EMAIL PROTECTED]
www.snowcrest.net


 How do I get the lights to work correctly on a SNOM 200 when I configure
it
 for more than one line?  The lights stay on solid, although the buttons
work
 correctly for making calls.  Thanks in advance.

 Dennis Engdahl
 SnowCrest, Inc.
 www.snowcrest.net

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Re: [Asterisk-Users] Linksys WRT54GS

2004-05-31 Thread TC

FYI the WRT54GS models have 8MB Flash  32MB RAM
and a 200Mhz Processor ~ about P166

- Original Message -
From: Tony Hoyle [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 31, 2004 11:37 AM
Subject: Re: [Asterisk-Users] Linksys


 Stefan de Konink wrote:
  Is Asterisk not a *little bit* too much for that processor? SER could be
a
  better choice?
 
 The asterisk binary alone is larger than the total flash ram space on the
linksys.

 I really doubt it's going to work

 Tony


 --
 Te audire no possum. Musa sapientum fixa est in aure.

 Tony Hoyle [EMAIL PROTECTED]  Key ID: 104D/4F4B6917 2003-09-13
 Fingerprint: 063C AFB4 3026 F724 0AA2  02B8 E547 470E 4F4B 6917
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[Asterisk-Users] Import Master.csv in the cdr_mysql database

2004-05-31 Thread Dan




Hi,

Has anyone succeed importing the Master.csv file in 
the cdr_mysql database using the tool import.php from the following 
address?
http://www.hotscripts.com/Detailed/29275.html
or there is any other available?

I have tried to use it, it say that the import was 
successfully, but ... nothing in the database.

Best regards,
Dan

RE: [Asterisk-Users] spandsp w/libtiff-3.6.1?

2004-05-31 Thread Sam Bingner

Did you actually look at that patch? --- It fixes some bug in 3.6.1
related to faxing... If so, sorry for wasting all your bandwidth :b

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Monday, May 31, 2004 1:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] spandsp w/libtiff-3.6.1?

Aaron J. Angel wrote:

 Has anyone used spandsp with a patched libtiff 3.6.1 successfully?

 http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500

Of course not. That is why I keep telling people not to use it. :-)

Regards,
Steve

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RE: [Asterisk-Users] spandsp wont compile.

2004-05-31 Thread Sam Bingner

You shouldn't put /usr/include in ld.so.conf, needing to do so means you
have something installed wrong... And I've never heard of anything getting
installed that wrong ;)

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone
Sent: Sunday, May 30, 2004 1:59 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] spandsp wont compile.


Yes, success! I deleted the tiff libs I had and installed ver 3.6.0 and
was able to compile and load the application modules. Now I just have to
do some tweaking and t-shootin' in ext.conf. Thanks and a Shout Out to all
for their advice and help. Couldn't have done it w/out you. I also had to
put /usr/include in ld.so.conf. Hope this helps others.


On Sat, 2004-05-29 at 18:09, Mark Musone wrote:
 Your most likely compiling against one tiff library version, but
 loading up another...

 Do a:

  ldd app_rxfax.so

 to see what tiff library it's compiled against,
 and then also try to find all the places where libtiff is on your
 machine and remove the incorrect one..

 -Mark


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Vlok Stone
 Sent: Saturday, May 29, 2004 6:09 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] spandsp wont compile.

 /etc/ld.so.conf

 /usr/X11R6/lib
 /usr/lib/qt3/lib
 /usr/local/libUnable to load module app_rxfax.so
 May 29 09:51:38 WARNING[1199209392]: loader.c:240 ast_load_resource:
 /usr/local/lib/libspandsp.so.0: undefined symbol: TIFFDefaultStripSize

 /usr/local/lib/libtiff
 /usr/lib/asterisk/modules

 the mods compiled BUT now won't load.

 On Fri, 2004-05-28 at 23:25, Todd Lieberman wrote:
  add /usr/local/lib to your /etc/ld.so.conf
 
  Then run ldconfig
 
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] Behalf Of Vlok
  Stone
  Sent: Friday, May 28, 2004 1:14 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] spandsp wont compile.
 
 
  got it to load but now it errors when starting asterisk. complains
  of
 no
  libspandsp.so.0 and its there. this fax thing is kickin my friggin
 fax!!
 
  On Fri, 2004-05-28 at 13:27, Vlok Stone wrote:
   I can't get spandsp to compile. when I go to the */apps directory
   i continually fails.
   Makefile:80: warning: overriding commands for target
   `app_rxfax.so'
   Makefile:77: warning: ignoring old commands for target
 `app_rxfax.so'
   cc -fPIC   -c -o app_rxfax.o app_rxfax.c
   app_rxfax.c:45: error: `PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP'
   undeclared here (not in a function)
   make: *** [app_rxfax.o] Error 1
  
   I chamged the Makefile to include
   app_rxfax.so : app_rxfax.o
   $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
  
   app_rxfax.so : app_rxfax.c
   gcc  -D_GNU_SOURCE  -O2 -g  -Iinclude  -l../include -c -o
   app_rxfax.   o app_rxfax.c
  
   app_txfax.so : app_txfax.o
   $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
  
   app_txfax.o: app_txfax.c
   gcc -D_GNU_SOURCE -O2 -g  -Iinclude -l../include -c -o
   app_txfax.o app_txfax.c
  
  
   any ideas?
   thanks in advance.
  
  
  
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[Asterisk-Users] SIP auth in/outside of nat?

2004-05-31 Thread Ben Hyde

I've got asterisk running behind a firewall.   On all machines in this 
narrative the domain name of the asterisk server resolves to the 
outside IP address.

The soft-phone can place calls just fine if it's outside the firewall.
Inside the firewall 'sip debug' that the soft-phone has connected but 
authentication appears to fail.

If I configure the soft-phone with the internal IP address of the 
asterisk server it works again.

I'm confused.  Help?
 - ben
softphone: x-lite/mac
asterisk 0.9.0 on freebsd
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RE: [Asterisk-Users] Re: Caller ID with BT CD50

2004-05-31 Thread Karl Dyson

Ok, I can report that I have just (within the last 2 hours, anyway)
downloaded the current cvs head for zaptel and asterisk, and applied
Tony's current patches downloaded freshly this evening from nodomain.
All applied and compiled, and with a tweak to my dring statements after
running asterisk in debug, cli and dring are both working :)

Calls with no CLI on either number go straight to voicemail without
ringing a phone (usually cold callers in my experience), and I lookup
the caller id in a mysql db using an AGI written in perl, that also
allows me to display which number the caller dialled as well as a
friendly name against the cli in the 7905 displays :)

Thanks All,

Karl

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Karl Dyson
 Sent: 30 May 2004 00:33
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Re: Caller ID with BT CD50
 
 Certainly Tony's original patch for CID works with my generic X101P
 (reports itself as an Intel 537 IIRC). I will get around to
downloading
 his new single patch that includes distinctive ringing and testing it
in
 the next couple of days.
 
 Cheers for now,
 
 Karl
 
  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Tony Mountifield
  Sent: 29 May 2004 19:06
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] Re: Caller ID with BT CD50
 
  In article [EMAIL PROTECTED],
  Tony Hoyle [EMAIL PROTECTED] wrote:
   Kevin Walsh wrote:
  
I downloaded the latest version of your patch, from your
website,
 and
it works perfectly.  I had waited until I had some time
available
because I thought I'd have to play around with it for a while.
  
   Great.   Just need to make sure that it still works for US lines
and
  it's all set.
  
   There's some debate whether to use this patch or to wait for one
 that
  uses
   line reversal/guard tone detection...   there is the slight
problem
 that
  the
   X100P can't detect line reversal so it'd mean everyone upgrading
 their
   hardware...  still, I have what 'works for me' and will continue
 hosting
  it
   for a while whatever happens.
 
  Is that the X100P generically, including the X101P?
 
  I would include both algorithms - the line reversal one for the
 hardware
  that can do it, and your current one for those that can't.
 
  Cheers,
  Tony
  --
  Tony Mountifield
  Work: [EMAIL PROTECTED] - http://www.softins.co.uk
  Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [Asterisk-Users] Galaxy Voice

2004-05-31 Thread Kevin

Thanks for your suggestion.  I will give it a try.  The other issue I
have is that the Galaxy service claims it has call waiting.  When one
call is up on the Galaxy connection, I get a busy when calling the
number, the same with an outbound, only one call at a time.

Thanks again,

Kevin


-Original Message-
From: Dr. Rich Murphey [mailto:[EMAIL PROTECTED] 
Sent: Monday, May 31, 2004 2:31 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Galaxy Voice

If it fails to register, check the sip debug output for:

REGISTER sip:216.229.127.40 SIP/2.0
Via: SIP/2.0/UDP 0.0.0.0:5060

If you see 0.0.0.0 in the 'Via' line, try using

nat=yes
externip=your external address

in your *global* section at the head of sip.conf.

I've searched but haven't been able to find where the value is being set
to
0.0.0.0.

Cheers,
Rich

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Kevin
 Sent: Saturday, May 29, 2004 1:24 PM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Galaxy Voice
 
 Yes, I did a search and have what I think is the correct
configuration.
 I did a google search and I didn't see much.  I was successful in
 getting it to work both inbound and outbound with the exception of the
 notices and warnings.
 
 The config I am using is:
 
 [galaxyvoice]
 nat=yes
 port=5060
 fromuser=12345678
 fromdomain=216.229.127.40
 username=12345678
 type=friend
 secret=12345678
 auth=md5
 host=216.229.127.40
 ;defaultip=216.229.127.40
 reinvite=no
 canreinvite=no
 dtmfmode=rfc2833
 context=inbound-galaxy
 qualify=yes
 disallow=all
 allow=gsm
 allow=ulaw
 callerid=12345678
 incominglimit=2
 defaultexpirey=60
 
 
 -Original Message-
 From: brian k. west [mailto:[EMAIL PROTECTED]
 Sent: Saturday, May 29, 2004 2:58 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Galaxy Voice
 
 Also I think someone posted a galaxy voice config example on the
mailing
 list a few weeks back.. have you searched google yet?
 
 bkw
 - Original Message -
 From: Kevin  [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, May 29, 2004 11:04 AM
 Subject: RE: [Asterisk-Users] Galaxy Voice
 
 
  I deeply apologize for the incorrect statement, thanks for taking
the
  time to point out the error...your help is appreciated.
 
  -Original Message-
  From: brian k. west [mailto:[EMAIL PROTECTED]
  Sent: Saturday, May 29, 2004 1:31 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Galaxy Voice
 
  First off they are not ERRORS  they are NOTICE and WARNING.
 
  bkw
 
  - Original Message -
  From: Kevin  [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Saturday, May 29, 2004 10:26 AM
  Subject: [Asterisk-Users] Galaxy Voice
 
 
   Has anyone successfully used Galaxy Voice with Asterisk?
  
   I am getting the following SIP errors repeated whether it is or
 isn't
   behind NAT.
  
   May 29 12:17:11 WARNING[1142135600]: chan_sip.c:595 retrans_pkt:
  Maximum
   retries exceeded on call
[EMAIL PROTECTED]
  for
   seqno 104 (Critical Request)
   May 29 12:17:25 NOTICE[1142135600]: chan_sip.c:3597
sip_reg_timeout:
   Registration for '[EMAIL PROTECTED]' timed out, trying
again
   May 29 12:22:52 WARNING[1142135600]: chan_sip.c:595 retrans_pkt:
  Maximum
   retries exceeded on call
[EMAIL PROTECTED]
  for
   seqno 111 (Critical Request)
   May 29 12:23:06 NOTICE[1142135600]: chan_sip.c:3597
sip_reg_timeout:
   Registration for '[EMAIL PROTECTED]' timed out, trying
again
   asterisk2*CLI
  
  
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Re: [Asterisk-Users] Snom and multiple lines

2004-05-31 Thread Rich Adamson

 So, first,
 why do the lights stay on, and secondly, can they light when anyone is using
 that extension?
snip
 Not only do we need the secretay/boss key system arrangement, but a
 traveling technician would like to be able to add his SIP extension to
 someone else's phone when he is working at their station.
 
snip
  How do I get the lights to work correctly on a SNOM 200 when I configure
 it
  for more than one line?  The lights stay on solid, although the buttons
 work
  correctly for making calls.  Thanks in advance.

I'm using a snom 200 v2.03o with two extns defined, and the lights work
as expected. (They didn't on some earlier version though.)
Make sure to define the two (or more) buttons in web interface under
Key Mappings (P1 = Line = Number sip:[EMAIL PROTECTED], P2 = ...),
matching Settings, SIP, Lines registered Account numbers.

If I press extn button #2 and place a call, the callerid properly indicates
the correct extension. If I call the extn number assigned to button #2,
the LED correctly flashes indicating an incoming call. When the call is
complete, all LEDs are off.

Regarding your key system question, I've never heard of anyone with a
configuration that would actual light button #2's LED if some remote
sip phone happened to be on the extension number assigned to that key.
If you could dream up a way to do that, it would be dependent on the
exact sip phone that you're using. There are no sip standards for
turning on/off LED's like that other then the MWI.

Regarding the traveling technician, certainly would not be all that difficult
to configure a dialplan that would provide that function. If the technician's
real extn was , have the technician dial a special extn from where ever
he happens to be (maybe x2111), and the code for that extension in the
dialplan implements a staight call forward to whatever the callerid happened 
to be for that call. (There's probably a dozen different ways to do that.
Check the wiki and goggle for examples.) Dialing x2111 again when he's
ready to leave that desk could also toggle the call forwarding off again.

Rich



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RE: [Asterisk-Users] D-Channel Problems

2004-05-31 Thread Joe Dennick

Like everyone else, I'm only guessing here.  I've got two different T-1s
running on two different Asterisk PBXs.  One of them is a PRI circuit
from ATT.  Getting that T-1 up was interested because of the cabling.
Is it possible that you've got a Cross-over cable connecting the
Asterisk to the D-Mark?  When dealing with this type of problem; its
best to start with the basics and work up.  If the wires are crossed; it
would impact the D-Channel.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff Sczpel
Sent: Monday, May 31, 2004 9:44 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] D-Channel Problems


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Good day eveyone,

I'm hoping that someone can help me.  Perhapps i'm overlooking the
obvious,  in truth, I hope that I am.  I've scoured the mailing list and
google,  and haven't come up with much.

I have a Digium T400P thats been connected to a channel bank for testing
for some time now.  all has been well.

I've now just had installed a PRI from Allegiance Telecom.  All 23
chanels are active, and they've told me that the 24th is the dchannel.
switchtype is Lucent 5ess.  Trunk is esf  b8zs

[/etc/zaptel.conf]
span=1,1,1,esf,b8zs
bchan=1-23
dchan=24
loadzone=us
defaultzone=us

[/etc/asterisk/zapata.conf]
[channels]
switchtype=5ess
context=incoming-pri
signalling=pri_cpe
echotraining=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
echocancel=yes
echocancelwhenbridged=no
group=5
immediate=no
callerid=asreceived
musiconhold=default
channel = 1-23

When I start Asterisk, I get this;
- 
May 30 21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event:
Alarm cleared on channel 10 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 11 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 12 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 13 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 14 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 15 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 16 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 17 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 18 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 19 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 20 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 21 May 30
21:54:07 NOTICE[1184048960]: chan_zap.c:5077 handle_init_event: Alarm
cleared on channel 22 May 30 21:54:07 NOTICE[1184048960]:
chan_zap.c:5077 handle_init_event: Alarm cleared on channel 23 May 30
21:54:07 WARNING[1175660480]: chan_zap.c:6174 zt_pri_error: PRI: Read on
130 failed: Unknown error 500 May 30 21:54:07 NOTICE[1175660480]:
chan_zap.c:6905 pri_dchannel: PRI got event: 5 on span 1
- --

and Asterisk never brings up the d-channel.  No calls are successful in
or out. (cannot create channel of type zap)


ANY ideas?
MUCH Appreciated if anyone can be of assistance.

Thanks,
Jeff
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Re: [Asterisk-Users] Re: Users in MySQL

2004-05-31 Thread Reto Stauss

 In the sip case it is the consequence of the mode MYSQL_FRIENDS is
 implemented. Probably the same with iax2.

Thanks for the clarification.

 When a sip phone registers, the current IP address and other
 parameters get updated in the database. This data will be used if
 there is a call to this phone.

Perfectly true. Because clients are behind NAT we use IAX. Unfortunately we get always
Rejected connect attempt and no authority found back. Any idea?

I double checked the name and secret in the db and the client configuration (using 
latest
firefly dev version).

Thanks!

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RE: [Asterisk-Users] Snom and multiple lines

2004-05-31 Thread Joe Dennick

For the Secretary/boss thing; you can also have inbound calls for the
boss ring on both the boss's phone and the secretary's phone.  Other
options include the AST-Gui-Client or the AST-Panel which are both
capable of showing who's on their phone at any given time.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Monday, May 31, 2004 5:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Snom and multiple lines

 So, first,
 why do the lights stay on, and secondly, can they light when anyone is

 using that extension?
snip
 Not only do we need the secretay/boss key system arrangement, but a 
 traveling technician would like to be able to add his SIP extension to

 someone else's phone when he is working at their station.

snip
  How do I get the lights to work correctly on a SNOM 200 when I 
  configure
 it
  for more than one line?  The lights stay on solid, although the 
  buttons
 work
  correctly for making calls.  Thanks in advance.

I'm using a snom 200 v2.03o with two extns defined, and the lights work
as expected. (They didn't on some earlier version though.) Make sure to
define the two (or more) buttons in web interface under Key Mappings (P1
= Line = Number sip:[EMAIL PROTECTED], P2 = ...), matching Settings,
SIP, Lines registered Account numbers.

If I press extn button #2 and place a call, the callerid properly
indicates the correct extension. If I call the extn number assigned to
button #2, the LED correctly flashes indicating an incoming call. When
the call is complete, all LEDs are off.

Regarding your key system question, I've never heard of anyone with a
configuration that would actual light button #2's LED if some remote sip
phone happened to be on the extension number assigned to that key. If
you could dream up a way to do that, it would be dependent on the exact
sip phone that you're using. There are no sip standards for turning
on/off LED's like that other then the MWI.

Regarding the traveling technician, certainly would not be all that
difficult to configure a dialplan that would provide that function. If
the technician's real extn was , have the technician dial a special
extn from where ever he happens to be (maybe x2111), and the code for
that extension in the dialplan implements a staight call forward to
whatever the callerid happened 
to be for that call. (There's probably a dozen different ways to do
that. Check the wiki and goggle for examples.) Dialing x2111 again when
he's ready to leave that desk could also toggle the call forwarding off
again.

Rich

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Re: [Asterisk-Users] line config

2004-05-31 Thread Nicholas Ruddick

Sean McKay wrote:
I was wondering, might it be possible to setup the following scenario:
SIP/iconnecthere - internal extension 1
SIP/bt - internal extension 2
..
Have it be transparent. internalext 1 would appear as my iconnect
phone number, as well as internalext 2 would appear as my bt number
so if I pick up extension2 it will only dial out to my bt line,
as with ext 1 would only dial out to iconnecthere. If someone rings
my bt line, ring extentsion 2, then voicemail(user 1000 i.e.), if
someone rings my iconnecthere line ring extension 1, then voicemail
(same voicemail user).
 

Yes, yes, yes and yes.
Now to add onto that. Would it be possible to hook both of those
2 extensions into 2 or more ip phones and have the ability to
put a call on hold on one phone while be able to pick it up
(same extension) on other phone.
yes
Have the ability to tell me
on one phone that another phone has a specified extension engaged
 

yes
and also be able to pick up and join in a current call without
call parking, or anything else like that.
 

Not sure, but probably.
thanks
sean
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[Asterisk-Users] MGCP RFC 3015

2004-05-31 Thread J C

Friends i am new to Asterisk.
I have bunch of Residential gateways Megaco RFC 3015 complaint.
Has any one been successful in installing and running Asterisk in production 
environment with RGW RFC 3015 complaint clients ??

I am excited to see my RGW come to play if i can get Asterisk to work.
Pros please advise me on Installation and Configuration of Asterisk to 
support Megaco Clients.

Thank you,
Java
_
Learn to simplify your finances and your life in Streamline Your Life from 
MSN Money. http://special.msn.com/money/0405streamline.armx

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[Asterisk-Users] RE: H323

2004-05-31 Thread T. Chan

Dear All,

I have used Asterisk for a few months and I have been using a January CVS
version, it has been working but has been regularly crashing. I use Asterisk
mostly as a softswitch application receiving H323 calls from customers and
send to H323 carriers. Since I have been using an older CVS version, the
OpenH323 and Pwlib libraries in use have been 1.11.7 and 1.4.11
respectively.

I was thinking of using a current asterisk version and see if it is more
stable comparing to the version in use. I upgraded to new version, and I
understand that with the new version and the H323 code, I need to use the
1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries respectively.
I have, therefore, erased the whole Pwlib and Openh323 folders, recreated
with the new versions and did the ./configure.make clean.make opt
procedures to compile the drivers.

I have then compiled all the zaptel, libpri, asterisk as usual, but when I
ran the asterisk, I noticed that most calls (calls mostly were sent from
Cisco AS5300 and Cisco AS5350) were getting one way audio, the calling party
was not able to hear anything even the call was connected, I am not sure if
the called party would hear anything, but obviously something is not working
properly.

Can any of your experts out there help please, thanks?

TC
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[Asterisk-Users] digium card fax detect AND spandsp

2004-05-31 Thread Ryan Courtnage

Hi all,
I've run into 2 separate problems relating to fax:
1) Using tdm400p + fxo, Asterisk is unable to detect the fax from some 
fax machines (from others it can).  Using zap barge, I can confirm that 
these troublesome calling fax machines _do_ send the fax tone loud and 
clear.  Are there certain circumstances where I should expect a Digium 
card to fail in detecting a fax?

2) Using spandsp  rxfax, I can consistently reproduce a problem where 
my client's * box cannot receive a fax sent from Windows-based HotFax.  
HotFax hangs up with error Phase B Error.
Watching the verbose * console, it appears that there is some useful 
communication between spandsp and the HotFax (output is here: 
http://voxbox.ca/tmp/rxfax.html).

Oddly enough, faxing between HotFax and spandsp _used_ to work, before 
I switched her * box over from an x100p to a tdm400p+fxo (although I 
can't be 100% certain the card change is to blame, since zaptel and 
asterisk have also both been updated).

Any comments on either of these issues is appreciated,
Thanks
RC
Ryan Courtnage
Coalescent Systems Inc
403.244.8089
www.voxbox.ca
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[Asterisk-Users] Firefly / LibIAX2

2004-05-31 Thread Reto Stauss

Hi

Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features 
when
using Firefly (Messaging, Status Indication).

The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not 
contain
any directions how to compile.

Any hints?

Thanks!
Reto

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[Asterisk-Users] Fwd: [Serusers] CDR mediation for VoIP

2004-05-31 Thread John Todd

FYI, for those of you who aren't on the serusers list.
I'd like to hear how others can get this working in small Asterisk 
settings; I don't really have the time to implement it, but it looks 
very interesting.

JT

To: [EMAIL PROTECTED]
From: Adrian Georgescu [EMAIL PROTECTED]
Date: Mon, 31 May 2004 23:05:47 +0200
Subject: [Serusers] CDR mediation for VoIP
For those of you who made inquiries about CDRTool (CDR mediation 
software for SER, Cisco and Asterisk),  I have made it available for 
download at:

http://www.ag-projects.com/OSS_CDRtool.html
CDRTool software is free to use for non comercial purposes.
Best regards,
Adrian Georgescu
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[Asterisk-Users] Interconnecting asterisk and SER configuration

2004-05-31 Thread Zouhair Echchelh

Hi,

can someone tell me how to intecrconnect an asterisk box with SER from
iptel.org

Thanks.
Zouhair Echchelh
Option-Service.fr

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Re: [Asterisk-Users] Snom and multiple lines

2004-05-31 Thread John Todd

At 4:05 PM -0600 on 5/31/04, Rich Adamson wrote:
  So, first,
 why do the lights stay on, and secondly, can they light when anyone is using
 that extension?
snip
 Not only do we need the secretay/boss key system arrangement, but a
 traveling technician would like to be able to add his SIP extension to
 someone else's phone when he is working at their station.
snip
  How do I get the lights to work correctly on a SNOM 200 when I configure
 it
  for more than one line?  The lights stay on solid, although the buttons
 work
  correctly for making calls.  Thanks in advance.
I'm using a snom 200 v2.03o with two extns defined, and the lights work
as expected. (They didn't on some earlier version though.)
Make sure to define the two (or more) buttons in web interface under
Key Mappings (P1 = Line = Number sip:[EMAIL PROTECTED], P2 = ...),
matching Settings, SIP, Lines registered Account numbers.
If I press extn button #2 and place a call, the callerid properly indicates
the correct extension. If I call the extn number assigned to button #2,
the LED correctly flashes indicating an incoming call. When the call is
complete, all LEDs are off.
Regarding your key system question, I've never heard of anyone with a
configuration that would actual light button #2's LED if some remote
sip phone happened to be on the extension number assigned to that key.
If you could dream up a way to do that, it would be dependent on the
exact sip phone that you're using. There are no sip standards for
turning on/off LED's like that other then the MWI.
[snip]
Rich

Actually, there do exist standards that would be able to provide the 
functions you're talking about with LED lighting based on who was on 
what extension.

The SIP SUBSCRIBE/NOTIFY tools were written to some degree with that 
type of feature in mind.  In fact, the rumor is that the limited 
SUBSCRIBE/NOTIFY support in Asterisk is specifically for the Snom 
phones, but I don't know (and doubt) if it does exactly this.

I _do_ know that Snom has touted that they are SUBSCRIBE/NOTIFY 
compliant, but I don't know the exact methodology of how they light 
up lights, put things on the LCD, or whatever.  If someone wants to 
send me a Snom 220, I'll be happy to figure out what's required.  :-)

The Polycoms are also rumored to support this type of feature, but 
again I don't know the exact mechanics to make it work.

JT
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Re: [Asterisk-Users] Snom and multiple lines

2004-05-31 Thread joachim

i think oej is working on something like that...
Zoa.
At 00:27 1/06/2004, you wrote:
At 4:05 PM -0600 on 5/31/04, Rich Adamson wrote:
  So, first,
 why do the lights stay on, and secondly, can they light when anyone is 
using
 that extension?
snip
 Not only do we need the secretay/boss key system arrangement, but a
 traveling technician would like to be able to add his SIP extension to
 someone else's phone when he is working at their station.
snip
  How do I get the lights to work correctly on a SNOM 200 when I configure
 it
  for more than one line?  The lights stay on solid, although the buttons
 work
  correctly for making calls.  Thanks in advance.
I'm using a snom 200 v2.03o with two extns defined, and the lights work
as expected. (They didn't on some earlier version though.)
Make sure to define the two (or more) buttons in web interface under
Key Mappings (P1 = Line = Number sip:[EMAIL PROTECTED], P2 = ...),
matching Settings, SIP, Lines registered Account numbers.
If I press extn button #2 and place a call, the callerid properly indicates
the correct extension. If I call the extn number assigned to button #2,
the LED correctly flashes indicating an incoming call. When the call is
complete, all LEDs are off.
Regarding your key system question, I've never heard of anyone with a
configuration that would actual light button #2's LED if some remote
sip phone happened to be on the extension number assigned to that key.
If you could dream up a way to do that, it would be dependent on the
exact sip phone that you're using. There are no sip standards for
turning on/off LED's like that other then the MWI.
[snip]
Rich

Actually, there do exist standards that would be able to provide the 
functions you're talking about with LED lighting based on who was on what 
extension.

The SIP SUBSCRIBE/NOTIFY tools were written to some degree with that type 
of feature in mind.  In fact, the rumor is that the limited 
SUBSCRIBE/NOTIFY support in Asterisk is specifically for the Snom phones, 
but I don't know (and doubt) if it does exactly this.

I _do_ know that Snom has touted that they are SUBSCRIBE/NOTIFY compliant, 
but I don't know the exact methodology of how they light up lights, put 
things on the LCD, or whatever.  If someone wants to send me a Snom 220, 
I'll be happy to figure out what's required.  :-)

The Polycoms are also rumored to support this type of feature, but again I 
don't know the exact mechanics to make it work.

JT
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Re: [Asterisk-Users] Unblocking incoming SIP

2004-05-31 Thread Duane

Brian Cuthie wrote:

Except that I *want* anyone to be able to call me directly from the 
Internet. That's the whole point -- we're trying to remove the necessity 
for a phone-company-like entity in the middle.

Instead, I suggest setting the default context for sip to something like 
sip-incoming-default and then include in the dialplan those things you 
wish people to be able to call directly.
Can we please get this option sooner then later... Needing to add a lot 
more config to overcome a problem that may or may not have been 
effecting most of us. Also even if they did have the config they still 
needed to add extension lines for or the call was still rejected!

--
Best regards,
 Duane
http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers
In the confrontation between the stream and the rock, the
stream always wins; not through strength, but through persistence.
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Re: [Asterisk-Users] Firefly / LibIAX2

It's the standard LibIAX2, the nice features are implemented using text 
messages. I'd recommend you use the standard LibIAX2 as it's more upto 
date (Something I've been needing to do too)

Reto Stauss wrote:
Hi
Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features 
when
using Firefly (Messaging, Status Indication).
The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not 
contain
any directions how to compile.
Any hints?
Thanks!
Reto
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[Asterisk-Users] wake-up call

2004-05-31 Thread Julian Pawlowski

Hi there!
I just try to play with die wake-up function described in
http://www.voip-info.org/wiki-Asterisk+tips+wake-up
Everything looks fine but there seem to be missing some soundfiles like 
wakeup-menu. Where can I get these files in order to make this feature 
usable?

Regards
Julian Pawlowski
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Re: [Asterisk-Users] spandsp w/libtiff-3.6.1?

2004-05-31 Thread Steve Underwood

Hi Sam,
I get many people complaining that spandsp is buggy because it crashes, 
when they simply haven't followed the instructions which say not to use 
3.6.1. Its getting really annoying, so I gave a rather sharp response. I 
did look at the patch, and perhaps my reply should have been a little 
more detailed. It is not clear to me whether that patch fixes all the 
problems. If someone would like to experiment with 3.6.1 using that 
patch, and report their results it would be helpful. If people want to 
just build and run spandsp, I would avoid 3.6.1, patched or otherwise, 
at this time.

Regards,
Steve

Sam Bingner wrote:
Did you actually look at that patch? --- It fixes some bug in 3.6.1
related to faxing... If so, sorry for wasting all your bandwidth :b
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Monday, May 31, 2004 1:53 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] spandsp w/libtiff-3.6.1?
Aaron J. Angel wrote:
 

Has anyone used spandsp with a patched libtiff 3.6.1 successfully?
http://bugs.hylafax.org/bugzilla/show_bug.cgi?id=500
   

Of course not. That is why I keep telling people not to use it. :-)
Regards,
Steve
 

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Re: [Asterisk-Users] Linksys

2004-05-31 Thread Stefan de Konink

On Mon, 31 May 2004, Greg Boehnlein wrote:

 On Mon, 31 May 2004, Tony Hoyle wrote:

  Stefan de Konink wrote:
   Is Asterisk not a *little bit* too much for that processor? SER could be a
   better choice?
 
  The asterisk binary alone is larger than the total flash ram space on the linksys.
 
  I really doubt it's going to work

 That assumes that compiling for the MIPS w/ uClibc is going to result in a
 binary that is similar in size to that of an x86 binary, which isn't
 neccessarily going to be true.

For your information * compiles in a clean uClibc env. expect for one or
two codecs, who use some x86 specific assembly. I have it on my system for
testing with an Epia.

Ok... you could run it with a remote NFS mount :-)


Stefan

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[Asterisk-Users] audio problems between asterisk and Cisco 7910 using SCCP

2004-05-31 Thread Mark Mills


Hi,

I am working with a friend to setup two Asterisk servers over the 
internet, one at each location and using IAX2 for trunking calls, using 
Cisco 7910 phones and chan_sccp.   The phones are all the same hardware 
and firmware revisions.

Lets call the sites AsteriskA and AsteriskB.   PhoneA is at AsteriskA, 
PhoneB is at AsteriskB.

PhoneA has problems, when calling the local voice mail service at
AsteriskA, the prompts are heard, button presses work, but audio does not
appear to reach the asterisk server.  The following error message appears
within the asterisk console:

Jun  1 08:43:01 WARNING[13326]: app_voicemail.c:1222 play_and_record: No 
audio available on SCCP/201-0001

The voice mail files that are created are empty.   Performing a packet 
dump I do see packets going to the Asterisk server.

Now also IAX2 is setup between AsteriskA and AsteriskB, and that seems to 
be functioning.   PhoneA and PhoneB can call each other from either 
direction, but once again there is no sound coming from PhoneA, its only 
one way.   If PhoneA is not answered, voicemail works and PhoneB can leave 
messages that PhoneA can retrieve, but not the other way around.

We performed a packet dump When making calls between the two locations, 
PhoneA sends data to AsteriskA, but AsteriskA doesnt forward it to 
AsteriskB.   It seems that the voice traffic is going from PhoneA is not 
being accepted at all?

Below is the config files that are in use for this setup. This has been 
compiled from source using asterisk-0.9.0.tar.gz and 
chan_sccp.02-easter.tar, on a Redhat 9 box running kernel 2.4.20-8. 

Does anyone have any idea what could be the problem and what we have 
missed?

Thanks,
  Mark


/etc/asterisk/sccp.conf
==
[general]

keepalive = 300
context = default
dateFormat = D/M/Y  

[SEP000427E8CD80]
type= 7910
autologin   = 201
description = Extension 201

[201]
id  = 201
pin = 1234
label   = Mark Mills 201
description = Mark Cisco 7910 Phone
callwaiting = 1
mailbox = 201
callerid= Mark Mills, 201




/etc/asterisk/extensions.conf
==
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for 
demo

[unknown]

exten = _.,1,Congestion

[default]

exten = 201,1,Macro(std-exten,SCCP/201,40)
exten = _1XX,1,Dial(IAX2/asterisk:[EMAIL PROTECTED]/[EMAIL PROTECTED]) 

exten = 999,1,wait(1)
exten = 999,2,VoicemailMain(${CALLERIDNUM})
exten = 999,3,Hangup

[macro-std-exten]
exten = s,1,Dial(${ARG1},${ARG2})
exten = s,2,Voicemail(u${MACRO_EXTEN})
exten = s,3,Hangup
exten = s,102,Voicemail(b${MACRO_EXTEN})
exten = s,103,Hangup






/etc/asterisk/modules.conf
==
[modules]
autoload=yes
noload = pbx_gtkconsole.so
noload = pbx_kdeconsole.so
noload = app_intercom.so
load = chan_modem.so
load = res_musiconhold.so
noload = chan_alsa.so
noload = chan_skinny.so
load = chan_sccp.so
noload = chan_oh323.so

[global]
chan_modem.so=yes



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Re: [Asterisk-Users] New Firefly version