RE: [Asterisk-Users] Pros and Cons of T1/E1 cards

2005-11-28 Thread David Waugh
Hi Steef,

Do you want to send me an email to [EMAIL PROTECTED] and I can assist you 
further.
It should work as far as I am aware.

Thanks David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of asterisk
Sent: 28 November 2005 07:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Pros and Cons of T1/E1 cards


Hi,

does any of you have experience with these cards on SMP 64bit systems?
I'm trying to get one to work but after fixing some some og the code 
they still do not work.
When installing single CPU and 32 bit they work ok.

Steef

David Waugh wrote:

Hi John,

I'm going to have to disagree with some previous posts.

The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the 
load of the call handling, echo cancellation etc as this is all processed on 
board on the card, and not on the central CPU of the computer.

You can use the CAPI interface of the card combined with chan_capi_cm with the 
card.
I have not found any problems when using different kernels or different 
versions of asterisk.
I have one setup in our test lab here at Eicon with Asterisk so it does work!

You can have up to 8 Diva Server cards in once machine - including a mixture 
of the analog and BRI cards.

The Diva Server cards in two variants - the V-Series if you only want to use 
them with Voice based applications and the normal All-in-one cards if you want 
to do fax and RAS too.

If you need any more information let me know, and I will assist further

David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Daragon
Sent: 25 November 2005 00:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Pros and Cons of T1/E1 cards


Hi;

We're looking to standardise on a single family of E1 PRI cards.

I guess our options are :

Digium / Zaptel / libpri
Sangoma/ Zaptel / Wanpipe
AVM/ CAPI
eIcon  / CAPI
Junghanns  / Bristuff

Can anyone share any comparative experience of these, please ? Do they 
differ much in terms of interrupt requirement, CPU load c ?

Any info gratefully received.

jd

  

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Re: [Asterisk-Users] Truncated CDR records

2005-11-28 Thread Leif Neland

 Original Message 
From: Innocent Evil [EMAIL PROTECTED]

you can use 'w' option with 'Dial' on 1.2.x



I don't think w do anything like 'wait', If I am wrong, correct me
someone please According to app_dial.c

w- Allow the called party to enable recording of the call by
sending\nthe DTMF sequence defined for one-touch
recording in features.conf.\n W- Allow the calling party to
enable recording of the call by sending\nthe DTMF
sequence defined for one-touch recording in features.conf.\n;

There is a difference between a w tagged on to the number, and a w as an 
option.

The options come last in the dial command, after a |

Leif

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Re: [Asterisk-Users] cdr enhancement with 'rate' column

2005-11-28 Thread Are
This is the purpose of AstBill at http://astbill.com

It gives you cost based CDR's out of the box.

AstBill is an Open Source Web Based Billing, Routing and Management
Software for Asterisk and VOIP. AstBill Provides pre and post paid
billing services and have a calling card module. AstBill completely
automates Asterisk and VOIP billing from start to finish. Key benefits
are the Central Web-based installation, Credit Control on outgoing and
incoming calls and the call routing module. Alternative you can use it
as a low cost office or home PABX.
-- Are Casillahttp://astartelecom.com - Independent VOIP Telecoms Broker. Asterisk Consultantshttp://astbill.com - Open Source Billing, Routing and Management software for Asterisk and VOIP
AstBill DEMO: http://demo.astbill.com
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[Asterisk-Users] a2billing / php agi debugging

2005-11-28 Thread Danish Samad
HI,



I have been playing around with the A2billing agi for a few days( I
really think its a great application) and trying to understand its
working. As far as I have understood, the only debugging mechanism is
going through the logs it generates. 

It would be really cool if I can run the agi in debug mode and
step through the code. I have been searching around and found nothing
related to debugging agi's (specifcally php agi's). If this is possible
can someone please tell me how it can be achieved. If its not possisble
please do tell me why and what are the alternates.



Regards,

Danish
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Re: [Asterisk-Users] A rather big setup.

2005-11-28 Thread Simone Cittadini

Vedran Dakic ha scritto:

 

 

How does Asterisk handle this kind of setup with one-two/cluster 
central server(s) and a bunch of other servers


connected with IAX(2)? If you have local calls, do they go directly 
from phone to phone, do they go from phone to


per-floor-Asterisk server, or they have to be interconnected via the 
main Asterisk server(s)/cluster?


With SIP the default is to directly connect the phones once the call is 
setup (I think also in IAX), investigate canreinvite / nat.


Of course you can't do call detail record for calls which aren't forced 
to pass from the server, see if it's a problem ...


imho
As for maintenance we have a dozen of pcs with asterisk installed, each 
of them is server for 8/10 sip phones and client to a central asterisk 
server which then connects to E1. Asterisk pcs are scattered around, 
they pass trough at least one natted network, usually two. Never a problem.


Connecting all the SIP phones to SER  load balancing to more than one 
asterisk server will make you learn a lot about sip internals, proxy, 
domains, authentication and other interesting stuff, but if you need to 
have a working sistems and can start from zero go iax and spare yourself 
a lot of frustration.

/imho
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Re: [Asterisk-Users] Dialplan pattern match discrepancy

2005-11-28 Thread Steve Davies
On 11/25/05, Daniel Wright [EMAIL PROTECTED] wrote:
 Steve Davies wrote:
  Hi,
 
  This is probably just me mis-reading the documentation, but I have
  been led to believe that the '.' in extensions.conf means zero or more
  digits, such that
 
  exten = _X.,1,NoOp()
 
  Would trigger for either a single digit, or for a longer number (as
  long as it starts with a digit)
 
  In practice (I am using 1.0.7 and 1.0.9) the '.' seems to match *one*
  or more digits, so in the above example, a single digit is not matched
  as expected.
 
  Is this correct? A bug? Fixed in 1.2 ;-) ?
 

 The period is match 1 or more characters(can be a number or letter).
 So in your example, you are saying first match a number 0-9, then match
 any one or more characters.

 Dan

Thanks Dan.

It does make the extensions.conf warning that '_.' should be replaced
with '_X.' a bit counter-intuitive then :) I completely understand why
'_.' is risky, but '_X.' is not the same thing at-all!

I assume that the only way to match one digit, followed by
zero-or-more alphanumeric is to have 2 sets of rules:

exten = _X,1,NoOp(One Digit)
exten = _X.,1,NoOp(One Digit and more stuff)

even if the same operation is to be carried out.

Ho hum!
Steve
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Re: [Asterisk-Users] Zaptel errors on Debian

2005-11-28 Thread Tzafrir Cohen
On Sun, Nov 27, 2005 at 05:22:39PM -0300, Rodrigo Campos wrote:
 On 11/27/05, Geotrix [EMAIL PROTECTED] wrote:
   Hello,
 
  I am trying to install zaptel wcfxo with X101.P board on Debian sarge 
  without success.
  (previously compiled and worked OK on Redhat kernel)
 and in debian ?
 for compiling it in debian i suggest you to install kernel-headers
 (tha same version of your running kernel), module-assistant and dpatch
 (install all with apt or aptitude) and zaptel and zaptel-source, of
 course :)

m-a (module-assistant) and dpatch are dependencies of zaptel-source.
(dpatch shouldn't really be, actually. Just because an incorrectly-laid
makefile)

 When i compile it i copy the /boot/config-running.kernel.version to
 /usr/src/kernel-headers-running.kernel.version/.config
 and then make cd /usr/src; m-a build zaptel this will generate a .deb in .
 then just dpkg -i name.of.the.deb.file.generated and reboot. Then
 you should can modprobe zaptel, etc...

shorter version: m-a a-i zaptel

I personally don't like the text-based interface and rather work with
the scriptable version:

  m-a -t -i a-i zaptel

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: Re: [Asterisk-Users] Zaptel errors on Debian

2005-11-28 Thread Tzafrir Cohen
On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote:
 
 Well, thanks, it might be great your package yet I would like to know how to 
 adapt.
 I wouldn't like to rewrite Debian neither Asterisk but is somebody able to 
 advice 
 how you define modules in zconfig.h or whatever ?
 Any tip ?
 Geo

Why would you need to define modules? The package builds wcfxo. 
What exactly do you try to do?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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RE: [Asterisk-Users] a2billing / php agi debugging

2005-11-28 Thread Steve Totaro
This is so simple, you are going to kick yourself.  Type agi debug in
the console.

Thanks,
Steve

 -Original Message-
 From: Danish Samad [mailto:[EMAIL PROTECTED]
 Sent: Monday, November 28, 2005 3:52 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] a2billing / php agi debugging
 
 HI,
 
 I have been playing around with the A2billing agi for a few days( I
really
 think its a great application) and trying to understand its working.
As
 far as I have understood, the only debugging mechanism is going
through
 the logs it generates.
  It would be really cool if I can run the agi in debug mode and step
 through the code. I have been searching around and found nothing
related
 to debugging agi's (specifcally php agi's). If this is possible can
 someone please tell me how it can be achieved. If its not possisble
please
 do tell me why and what are the alternates.
 
 Regards,
 Danish
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Re: Re: [Asterisk-Users] Intel G729 Codec Install error on [EMAIL PROTECTED]

2005-11-28 Thread Wilson
Dear Tzafrir Cohen,

I paste the detail message for you,please see the below message. 
I wonder it may cause by different linux system.

[EMAIL PROTECTED] speech-coding]# patch -p1  /tmp/ipp-050903.diff
patching file G723.1/Makefile
patching file G723.1/samples/codec_g723.c
patching file G723.1/samples/g723_slin_ex.h
patching file G723.1/samples/print_bytes.c
patching file G723.1/samples/slin_g723_ex.h
patching file G729-float/Makefile
patching file G729-float/samples/codec_g729.c
patching file G729-float/samples/decoder.c
patching file G729-float/samples/encoder.c
patching file G729-float/samples/g729_slin_ex.h
patching file G729-float/samples/gen_test.c
patching file G729-float/samples/my_dec.c
patching file G729-float/samples/my_enc.c
patching file G729-float/samples/slin_g729_ex.h
patching file G729-float/samples/util_e.c
patching file G729-float/samples/util_e.h
patching file README.Asterisk
patching file debian/asterisk-ipp-codecs.dirs
patching file debian/asterisk-ipp-codecs.install
patching file debian/changelog
patching file debian/control
patching file debian/rules
patching file runme.sh
[EMAIL PROTECTED] speech-coding]# chmod a+x runme.sh
[EMAIL PROTECTED] speech-coding]# ls
AMRWB   G722.1  G728  G729-float  ipplic.htm   runme.sh
debian  G723.1  G729  GSMAMR  README.Asterisk  support.txt
[EMAIL PROTECTED] speech-coding]# ./runme.sh
Makefile:126: target `samples/util_e.o' given more than once in the same rule.
Makefile:126: target `samples/util_e.o' given more than once in the same rule.
Makefile:126: target `samples/util_d.o' given more than once in the same rule.
Makefile:126: target `samples/util_d.o' given more than once in the same rule.
rm -f  ./api/decg729fp.o  ./api/encg729fp.o  ./api/owng729fp.o  
./api/usc729fp.o  

  ./vm/src/vm_thread_linux32.o  ./samples/util_e.o  ./samples/util_d.o  
./sample  

s/codec_g729.o  ./samples/encoder.o  ./samples/util_e.o  ./samples/util_e.o  
./s  

amples/my_enc.o  ./samples/decoder.o  ./samples/util_d.o  ./samples/util_d.o  
./  

samples/my_dec.o
Makefile:126: target `samples/util_e.o' given more than once in the same rule.
Makefile:126: target `samples/util_e.o' given more than once in the same rule.
Makefile:126: target `samples/util_d.o' given more than once in the same rule.
Makefile:126: target `samples/util_d.o' given more than once in the same rule.
gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include 
  

/opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux 
-Dlinu  

x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 
-mcpu=pen  

tium4 -ffast-math -fomit-frame-pointer  -osamples/util_e.o samples/util_e.c
gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include 
  

/opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux 
-Dlinu  

x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 
-mcpu=pen  

tium4 -ffast-math -fomit-frame-pointer  -osamples/util_d.o samples/util_d.c
gcc -I./include -I./vm/include -I/opt/intel/ipp41/ia32_itanium/include -include 
  

/opt/intel/ipp41/ia32_itanium/tools/staticlib/ipp_w7.h -D__unix__ -Dlinux 
-Dlinu  

x32 -DNDEBUG -DLINUX32 -DNO_SCRATCH_MEMORY_USED -c -O6 -march=pentium4 
-mcpu=pen  

tium4 -ffast-math -fomit-frame-pointer  -osamples/codec_g729.o 
samples/codec_g72  

9.c
In file included from ./vm/include/vm_types.h:17,
 from ./vm/include/vm_thread.h:14,
 from samples/encoder.h:28,
 from samples/codec_g729.c:39:
./vm/include/sys/vm_types_linux32.h:36: error: syntax error before 
use_ast_cond  

_t_instead_of_pthread_cond_t
./vm/include/sys/vm_types_linux32.h:36: warning: no semicolon at end of struct 
o  

r union
./vm/include/sys/vm_types_linux32.h:37: warning: data definition has no type or 
  

storage class
./vm/include/sys/vm_types_linux32.h:40: error: syntax error before '}' token
./vm/include/sys/vm_types_linux32.h:40: warning: data definition has no type or 
  

storage class
./vm/include/sys/vm_types_linux32.h:51: error: syntax error before 
use_ast_mute  

x_t_instead_of_pthread_mutex_t

[Asterisk-Users] Problem with ADIT 600 and FXO configuration

2005-11-28 Thread William K. Volkman
I've looked through the archives of the mailing list for the
last year and although informative I've not been successful
at get this to work.  We had a working Asterisk PBX system
with 3 Digium X101P FXO lines and two TDM400P FXS cards.
I've setup an ADIT 600 with an 8 port FXO card (and an
8 port FXS card not currently installed).  We are going
to be adding a T1 for incoming calls this week. I removed
two of the X101P cards and installed a TE406P.  I'm using
Asterisk 1.0.9 (and matching zaptel, libpri) from tar files.

/etc/zaptel.conf has this configuration:
span=1,1,0,esf,b8zs,yellow
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
#Modular unit, first card is FXO
fxsks=1-3
unused=4-8
#Modular unit, 1 FXS cards
unused=9-16
unused=17-24
unused=25-48,49-72,73-96
fxsks=97
fxoks=98-101
fxoks=102-105

/etc/asterisk/zapata.conf has this:
group = 0
signalling=fxs_ks
context = incoming
busydetect = yes
overlapdial = no
channel = 1-3

signalling=fxs_ks
channel = 97  ;X100P 

group = 1
signalling = fxo_ks
context = internal
;TDM400P
callerid = Available 200
channel = 98-100
callerid = x
channel = 101
;TDM400P
callerid = x
channel = 102
callerid = x
channel = 103

Parts of my adit configuration:
-Setting slot a.
 
set a:1 up
set a:1 fdl none
set a:1 lbo 4
set a:1 framing esf
set a:1 id Inbound
set a:1 linecode b8zs
set a:1 loopdetect csu
set a:1:1-24 side drop
set a:1:1-24 type voice
set a:1:1-24 signal ls
set a:2 up
set a:2 fdl none
set a:2 lbo 1
set a:2 framing esf
set a:2 id Outbound PBX
set a:2 linecode b8zs
set a:2 loopdetect csu
set a:2:1-24 side drop
set a:2:1-24 type voice
set a:2:1-24 signal ls
   
-Setting slot 1.
 
set 1:1-8 signal lscpd
set 1:1-8 txgain -3
set 1:1-8 rxgain -6

-Setting primary and secondary clock sources.
  
set clock1 a:1
set clock2 internal

-Setting the system idle pattern for DS0s.
   
set idle 0xff
  
-Making connections.
 
connect a:2:1-3 1:1-3
  
Inbound calls just ring and ring (the leds on the ADIT change
state) however asterisk doesn't respond.  Attempts to make
outgoing calls get:
-- Executing Dial(SIP/202-ba07, Zap/g0/5551212) in new stack
Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create
channel of type 'Zap'
  == Everyone is busy/congested at this time
-- Executing Congestion(SIP/202-ba07, ) in new stack
  == Spawn extension (from-sip, 95942060, 3) exited non-zero on
'SIP/202-ba07'
-- Executing Hangup(SIP/202-ba07, ) in new stack

I've tried just about all combinations of gs/ls/ks for the
signalling to no avail.  Here is the output of status:

 status a:2:1-3
 
DS0 Rx AB  Tx AB  Signal  T1 TP
--- -  -  --  -  --
a:2:1 01 01  LS   TrafficN
a:2:2 01 01  LS   TrafficN
a:2:3 01 01  LS   TrafficN

 status 1:1-3
   
FXORx AB  Tx AB  Signal=T1 Sig  T1 TP
----  -  --  -  --
1:1  01 01   LSCPD = LS Traffic N
1:2  01 01   LSCPD = LS Traffic N
1:3  01 01   LSCPD = LS Traffic N
   
 show connect a:2:1-3
FromDescDescTo
 ---  --  -  -
  A:02:01LS VOICE   DS0 -- FXOVOICE LSCPD   1:01
  A:02:02LS VOICE   DS0 -- FXOVOICE LSCPD   1:02
  A:02:03LS VOICE   DS0 -- FXOVOICE LSCPD   1:03

Can anyone spot what I've got wrong?  Any suggestions or hints
welcome.

Thanks,
William.


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Re: [Asterisk-Users] a2billing / php agi debugging

2005-11-28 Thread Danish Samad
HI Steve,

Thanks for the reply.
I have tried agi debug before. Actually by debugging I mean
inserting breakpoints and stepping through the code. I have not been
able to find any solution for this sort of debugging setup.

Regards,
DanishOn 11/28/05, Steve Totaro [EMAIL PROTECTED] wrote:
This is so simple, you are going to kick yourself.Type agi debug inthe console.Thanks,Steve -Original Message- From: Danish Samad [mailto:
[EMAIL PROTECTED]] Sent: Monday, November 28, 2005 3:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] a2billing / php agi debugging
 HI, I have been playing around with the A2billing agi for a few days( Ireally think its a great application) and trying to understand its working.As far as I have understood, the only debugging mechanism is going
through the logs it generates.It would be really cool if I can run the agi in debug mode and step through the code. I have been searching around and found nothingrelated to debugging agi's (specifcally php agi's). If this is possible can
 someone please tell me how it can be achieved. If its not possisbleplease do tell me why and what are the alternates. Regards, Danish___
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[Asterisk-Users] Legacy PBX integration problem

2005-11-28 Thread Dmitry Kupchinetsky

Hi,
We are trying to integrate Asterisk in front of our existing legacy PBX:
outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions
Asterisk answers outside calls and the IVR asks the user to dial extension #.
The problem is, that when Asterisk forwards calls from the outside line 
to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, the call 
is answered. Asterisk then dials the extension number, but even if the extension 
is busy or no one answers, the call is answered as far as * is concerned.
We want Asterisk to:
-DIAL the PBX
-wait for the PBX to answer
-dial extension number
-If there is no answer at this extension start voicemail.
Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already answered the call?
Thanks,Kupchinetsky Dmitry

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[Asterisk-Users] Legacy PBX integration problem

2005-11-28 Thread Dmitry Kupchinetsky



Hi,
We are trying to integrate Asterisk in front of our existing legacy PBX:
outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions
Asterisk answers outside calls and the IVR asks the user to dial extension #.
The problem is, that when Asterisk forwards calls from the outside line 
to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, the call 
is answered. Asterisk then dials the extension number, but even if the extension 
is busy or no one answers, the call is answered as far as * is concerned.
We want Asterisk to:
-DIAL the PBX
-wait for the PBX to answer
-dial extension number
-If there is no answer at this extension start voicemail.
Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already answered the call?
Thanks,Kupchinetsky Dmitry

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[Asterisk-Users] Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread Mark van Kerkwyk

Hi, anyone managed to get a Presence
Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe
this should be paritally supported now in 1.2 ?

regards

Mark___
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[Asterisk-Users] Legacy PBX integration problem

2005-11-28 Thread Dmitry Kupchinetsky


Hi,

We are trying to integrate Asterisk in front of our existing legacy PBX:

outside line -- FXO -- Asterisk PC -- FXS -- PBX -- Extensions

Asterisk answers outside calls and the IVR asks the user to dial extension 
#.

The problem is, that when Asterisk forwards calls from the outside line
to the old PBX (using DIAL()) the PBX answers, and as far as * is concerned, 
the call
is answered. Asterisk then dials the extension number, but even if the 
extension

is busy or no one answers, the call is answered as far as * is concerned.

We want Asterisk to:
-DIAL the PBX
-wait for the PBX to answer
-dial extension number
-If there is no answer at this extension start voicemail.

Any ideas?
How can we detect when an extension is busy/no answer AFTER PBX already 
answered the call?


Thanks,

Kupchinetsky Dmitry


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RE: [Asterisk-Users] A rather big setup.

2005-11-28 Thread Roger Workman
Vedran,

Email me off topic and I can provide you some case studies of different 
providers for your review.

[EMAIL PROTECTED]

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Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
 Fax:   304.324.3801
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 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Jan Saell
 Sent: Monday, November 28, 2005 12:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] A rather big setup.

 if this is a brand new thing you can force the phones on people and then
 you can to provisioning remotly of for instance Grandstream so they can
 change the config themself. By forcing a common set of codex you can avoid
 cpu overhead of translation so you only have to think of teh datashuffle.

 Bu doing god work at the dialpla you make shure that all the calls thats
 internal never hit the main pbx'es in the celler and oly use them for
 outgoing!

 Best regards
 jan

 --On Monday, November 28, 2005 04:22:09 AM 퍝 Vedran Dakic
 [EMAIL PROTECTED] wrote:

 
  Hello,
 
  Those people currently aren't using any kind of phones, but the
 investment
  company that has this building in the works wants to deliver
 everything
  for them so they just have to - move in and do business.
 
  What worries me is the fact that when you have 100-200 offices - they're
  used to having 2-3 lines only for them - one for fax, two for voice,
 etc.
  So, in a way, having in mind around 200-300 outbound calls at peak time
 is
  pretty much normal. Also, when you think of the number of phones - it
  would only be normal to assume for people to have up to 1000 internal
  phone conversations peak (the less transcoding - the better, of course).
 
  I have a freedom of making whatever I want, so I can have a separate LAN
  for VoIP purposes only - a bunch of dedicated patch panels, VLANs on
 Cisco
  switches, or whatever. I'm just considering this setup way before it has
  to go online because of the price of traditional PBX for this kind of
  setup which can only make you hurl. And you know how much potential
  upgrades cost for a setup like this - a traditional PBX can be a
  nightmare :(
 
  Cheers,
  Vedran.
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Hans
  Witvliet Sent: Monday, November 28, 2005 12:08 AM
  To: asterisk-users@lists.digium.com
  Subject: RE: [Asterisk-Users] A rather big setup.
 
  I think there is more to consider.
  One or two fat machines in the basement forr connecting to the PSTN is
  very fine.
  But are all the people allready using voip handsets, or old fashioned
  analoge handsets? If so, you need quite a large number of channelbanks.
  You speak of 300/1500 concurrent phone calls? If so how many handsets
  are you considering?
  Is the lan capable of handling this load?
  Is the lan 100% dedicated for voip, or are there a bunch of
  servers/workstations also using this lan?
 
  Interesting project
 
  Hans
 
 
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 --
 +--
 ! Irial / YASK AB
 ! Att: Jan Saell
 ! Box 59, S-692 21 KUMLA, SWEDEN
 ! Tel: 019-58 25 15 Int 19 58 25 15 Fax 19 58 38 05
 ! E-mail: [EMAIL PROTECTED]
 ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD  7F18 404A 5DA1 F944 A08B



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RE: [Asterisk-Users] Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread harry gaillac
Don't waste your time asterisk does not support
presence
--- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :

 Hi, anyone managed to get a Presence Agent
 configuration with Asterisk 1.2 
 and X-Ten Eyebeam working. I believe this should be
 paritally supported 
 now in 1.2 ?
 
 regards
 
 Mark
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[Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Noc Phibee

Hi

it's possible to upgrade the firmware of a cisco 7910 with asterisk ?

he have a other solution for upgrade it without callmanager ?

thansk for your help

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[Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread Ben Buxton

Are you sure? I've got it working with Eyebeam, showing me just who is
available and who isn't. 

http://www.voip-info.org/wiki-Asterisk+phone+snom

A couple of pages down you'll see this:

 SNOM SUBSCRIBE/NOTIFY support for monitoring extension states 

The methods and configuration here are also valid for Eyebeam.

BB

harry gaillac [EMAIL PROTECTED] uttered the following thing:
 Don't waste your time asterisk does not support
 presence
 --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
 
  Hi, anyone managed to get a Presence Agent
  configuration with Asterisk 1.2 
  and X-Ten Eyebeam working. I believe this should be
  paritally supported 
  now in 1.2 ?
  
  regards
  
  Mark
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Sergio Chersovani

Noc Phibee ha scritto:


it's possible to upgrade the firmware of a cisco 7910 with asterisk ?


You need the legal firmware upgrade file
download the chan_sccp code from http://chan-sccp.berlios.de
configure it and use the imageversion param to upgradde the phone firmware.

Of course you need a tftpserver and if you run a tftpserver you just 
need a SEPmac to upgrade the phone

So the correct answer is:
you don't need a CCM nor asterisk to upgrade a cisco phone firmware.
You just need the firmware file, a tftpserver and a configuration file 
(SEPmac)


take a look here
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx


Sergio
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RE: [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread harry gaillac
I'm sure look at rfc3265 (SUBSCRIBE/NOTIFY) which is
not support by asterisk.
How can you monitor the states of the buddies ?

Harry
--- Ben Buxton [EMAIL PROTECTED] a écrit :

 
 Are you sure? I've got it working with Eyebeam,
 showing me just who is
 available and who isn't. 
 
 http://www.voip-info.org/wiki-Asterisk+phone+snom
 
 A couple of pages down you'll see this:
 
  SNOM SUBSCRIBE/NOTIFY support for monitoring
 extension states 
 
 The methods and configuration here are also valid
 for Eyebeam.
 
 BB
 
 harry gaillac [EMAIL PROTECTED] uttered the
 following thing:
  Don't waste your time asterisk does not support
  presence
  --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
  
   Hi, anyone managed to get a Presence Agent
   configuration with Asterisk 1.2 
   and X-Ten Eyebeam working. I believe this should
 be
   paritally supported 
   now in 1.2 ?
   
   regards
   
   Mark
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Re: [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread Mark van Kerkwyk

aha, it was the subscribecontext= that
was missing. Basic presence works fine now. Offline, online, on the phone
:-)

What about IM, did you have that working
too ?

thanks

Mark






Ben Buxton [EMAIL PROTECTED]

Sent by: [EMAIL PROTECTED]
28/11/2005 11:59 PM



Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com





To
asterisk-users@lists.digium.com


cc



Subject
[Asterisk-Users] Re: Presence + Eyebeam
+ Asterisk 1.2









Are you sure? I've got it working with Eyebeam, showing me just who is
available and who isn't. 

http://www.voip-info.org/wiki-Asterisk+phone+snom

A couple of pages down you'll see this:

 SNOM SUBSCRIBE/NOTIFY support for monitoring extension states 

The methods and configuration here are also valid for Eyebeam.

BB

harry gaillac [EMAIL PROTECTED] uttered the following thing:
 Don't waste your time asterisk does not support
 presence
 --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
 
  Hi, anyone managed to get a Presence Agent
  configuration with Asterisk 1.2 
  and X-Ten Eyebeam working. I believe this should be
  paritally supported 
  now in 1.2 ?
  
  regards
  
  Mark
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[Asterisk-Users] Re: Re: Presence + Eyebeam + Asterisk 1.2

2005-11-28 Thread Ben Buxton

Can't say I've actually tried IM, but Ill give it a go sometime. I think
the wiki needs updating on all this...the eyebeam page is very
incomplete on subscribe, im, etc.

BB

Mark van Kerkwyk [EMAIL PROTECTED] uttered the following thing:
 aha, it was the subscribecontext= that was missing. Basic presence works 
 fine now. Offline, online, on the phone :-)
 
 What about IM, did you have that working too ?
 
 thanks
 
 Mark
 
 
 
 
 Ben Buxton [EMAIL PROTECTED] 
 Sent by: [EMAIL PROTECTED]
 28/11/2005 11:59 PM
 Please respond to
 Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 
 
 To
 asterisk-users@lists.digium.com
 cc
 
 Subject
 [Asterisk-Users] Re: Presence + Eyebeam + Asterisk 1.2
 
 
 
 
 
 
 
 Are you sure? I've got it working with Eyebeam, showing me just who is
 available and who isn't. 
 
 http://www.voip-info.org/wiki-Asterisk+phone+snom
 
 A couple of pages down you'll see this:
 
  SNOM SUBSCRIBE/NOTIFY support for monitoring extension states 
 
 The methods and configuration here are also valid for Eyebeam.
 
 BB
 
 harry gaillac [EMAIL PROTECTED] uttered the following thing:
  Don't waste your time asterisk does not support
  presence
  --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit :
  
   Hi, anyone managed to get a Presence Agent
   configuration with Asterisk 1.2 
   and X-Ten Eyebeam working. I believe this should be
   paritally supported 
   now in 1.2 ?
   
   regards
   
   Mark
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 Messenger 
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[Asterisk-Users] troubles with voicemail

2005-11-28 Thread Matthias Leeb
hi list!

I've configured some voicemailboxes and at the beginning everything was
working fine. In the past few days, evetime i want to hear the messages,
recorded on my box the following lines come up the asterisk logfile:

Nov 28 14:12:31 WARNING[25446]: file.c:508 ast_openstream_full: File
digits/1F does not exist in any format Nov 28 14:12:31 WARNING[25446]:
file.c:820 ast_streamfile: Unable to open digits/1F (format ulaw): No such
file or directory
  == Spawn extension (sip, 82103, 1) exited non-zero on 'SIP/soft_leeb-273f'
Nov 28 14:12:31 ERROR[25446]: cdr_custom.c:127 custom_log: Unable to re-open
master file /var/log/asterisk/cdr-custom/Master.csv : No such file or
directory

Can anybody imagine why this error occurs?

Best regard,

Matthias

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Re: [Asterisk-Users] stop asterisk when Idle

2005-11-28 Thread asterisk
Thank you for your asnswer

I found that between 7 and 8 in the mornig I have a low load of my box.

I modify my script in this way:

asterisk02:/ # cat /closeasteriskandreboot.sh
#!/bin/bash
echo chiusura schedulata when convenient di asterisk
/usr/sbin/asterisk -rx stop when convenient

mypid=$(pidof asterisk)
echo mypid=${mypid}

while [ ${mypid}empty != empty ] ; do
  echo Il pid di asterisk esiste ancora ( = ${mypid} ) , aspetto altri 10
secondi
# adesso dormo per dieci secondi per aspettare che asterisk chiuda tutto
bene
  sleep 10
  mypid=$(pidof asterisk)
  echo mypid=${mypid}
done

echo Il pid di asterisk non esiste piu' , riavvio il pc
echo reboot del server, ciao ciao
/sbin/reboot

exit

***
I think that it should be possible to reach a convenient (idle) time.
I could also implement a sort of timeout (if in half an hour asterisk is
still up, then send stop now command and then reboot)...

On the other side, I didn'i install any monitoring utility on my box (I
don't konw them) The bigger problem is that it is a productivity box, and
to do a lot of experiments is not so easy

So I think that I will leave my box as it is now, and I will check in the
future if this strange oh323 problem will be found elsewhere and resolved.

thanks all,

Andrea



   
 Leif Neland 
 [EMAIL PROTECTED] 
 Sent by:   To 
 asterisk-users-bo Asterisk Users Mailing List -  
 [EMAIL PROTECTED] Non-Commercial Discussion  
 m.com asterisk-users@lists.digium.com   
cc 
   
 26/11/2005 17.35  Subject 
   Re: [Asterisk-Users] stop asterisk  
   when Idle   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   




 I still continue to reboot my asterisk box everyday.

 I posted a message on November 22, but it was on another thread and
 no one answered me, so I try again here,
 where a lot of people told be I was a bad administrator (Like a
 Windows administrator and I don'0t want to resolve my problem)

 Actually I would like to resolve my problem, but I am not able to do
 this, so I ask help to anybody who can help me, and repost my
 last of 22/11/2005

 In short, my problem is that, after one or two days of running, chan
 oh323 suddendly disappear from asterisk box, without giving any
 warning / error In example, you type oh323 show stats at 11 o'clock
 , and get an answer from asterisk, about usage of oh323

 At 12, without doing anything to the box or to the asterisk, you
 type the same command, and you get a  No such command 'oh323' (type
 'help' for help)

 If you type help, no oh323 commands are available.
 If you quit asterisk, (STOP NOW) and restart asterisk , no oh323
 channel command is available

 if you reboot the machine everything is again fine !

 It is so a crazy situation that to reboot appears (to me) the
 best thing (I
 am sorry about this)

If you really need to have oh323, then you should test say every 5 minutes
or so, and then shutdown asterisk and reboot.

if asterisk -r -x oh323|grep help
then
  echo oh323 missing|mail administrator
  asterisk -r -x stop now
  reboot
fi

Ii is better to disconnect the existing users if they can not use the box
without oh323

Do you have some kind of monitoring running? Like Big Brother or nagios?
It might be interesting to see when oh323 dies.
Perhaps you could also use mrtg to graph usage levels to see if there is
some kind of correlation between usage and oh323 fatality

Leif

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Re: [Asterisk-Users] Sangoma problems!?

2005-11-28 Thread Michael Bielicki
hat is the content of your wanpipe?.conf files ?On 11/26/05, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Stig Even Larsen wrote: I'm having problems connecting my Sangoma cards to our PRI (E1)
 interface. It seems that the card get connected (green led), but Asterisk reports: Status: Provisioned, Down, Active. When I install another Sangoma card on the same system (pri_net), and
 connect the two cards with a PRI cross-over cable both cards get connected (green led) and Asterisk reports both spans: Status: Provisioned, UP, ActiveYour telco has not brought the D-channel on your PRI circuit to 'up'
state yet. Plug the card back in, make the red alarm go away, and thencall them and tell them you are ready to use the circuit.___--Bandwidth and Colocation provided by 
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-- Michal BielickiHalo Kwadrat Sp. z o.o.http://www.asterisk.pl/http://www.openpbx.org/

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Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax

2005-11-28 Thread James Armstrong
I looked into why I can't get the original DID number called when a fax 
is detected (so I can later route to the correct email address). There 
is a variable called FAXEXTEN that is created when a fax is detected, 
but it is not being populated with the original extension / did number 
called. It always has 's' as the original extension. Does anyone know 
how to fix this? I just want to use the zap fax detection so each person 
can have their personal DID number accept faxes and calls, then route 
the fax to their email address based on the DID number called.


- James


James Armstrong wrote:

In this example faxdetect is overwriting the DID.

So the trick then would be to somewhere early-on in your dialplan grab 
the DID into some variable, and then restore it after the fax 
detection occurs...


[default]
exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN})
exten = s,2,Wait(3)
.
exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN})

Lee.



I am having no luck here. It seems the fax detection is overriding 
everything. I added the above and it never gets called. I guess it might 
be time to look at the Asterisk code and see if I can create another 
variable before the redirect happens.


-- Starting simple switch on 'Zap/1-1'
-- Redirecting Zap/1-1 to fax extension
-- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack

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[Asterisk-Users] Problem forwarding zap to sip

2005-11-28 Thread Miloš Kocbek
I have a problem when i have asterisk connected to voxtream parlay i60
PRA port.
When call is received to asterisk and forwarded to SIP IP gsm gateway
call is always disconnected with cause 102.

# 102 Recovery on timer expiry

This cause indicates that a procedure has been initiated by the expiry
of a timer in association with ETS 300 102-1 error handling
procedures.

Call is not disconnected if user answers the call in this 18 seconds.
I think the problem is that all messages from ip gateway don't get
forwardet to PRI channel and call is disconnected.

What can be done to solve this problem

greetings

mk
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RE: [Asterisk-Users] Sangoma problems!?

2005-11-28 Thread David Yat Sin
Hi Stig,
When I install another Sangoma card on the same system (pri_net), and
connect the two cards with a PRI cross-over cable both cards get connected
(green led) and Asterisk reports both spans:
Status: Provisioned, UP, Active

It does look your installation is fine, i.e you are being able to
send/receive packets. One way to confirm this is to perform a trace on span
1: (with the E1 cross-over cable between the two ports) 

1. In the Asterisk CLI, type:
pri intense debug span 1. 
You should see that packets are being transmitted in both directions, i.e.
Some packets will have 




On the left hand side and others will have   

2. In another command window (linux bash), type: 
Wanpipemon -I w1g1 -c trd

This is a sangoma tool that will trace the d-channel of span 1. Check if you
are having both incoming and outgoing packets. 

If for both tests, you are receiving and transmitting packets, this means
your connection is fine, but your telco line is not provisioned yet. Contact
your telco and tell them that you are not receiving packets. 

If you are receiving and transmitting packets from the wanpipemon trace, but
only transmitting or only receiving in the asterisk CLI, then this that the
Sangoma card is not interfacing properly with zaptel. This usually happens
because you did not re-compile zaptel after installing wanpipe. Please
re-compile zaptel. The correct installation sequence is:

1. Zaptel, Libpri, Asterisk
2. wanpipe
3. recompile zaptel. 
4. perform configurations (wancfg, zaptel.conf etc.. )

Do not hesitate to contact me if you have any questions. 


Regards,

David Yat Sin
Sangoma Technologies
(905) 474-1990 x119
(800) 388-2475 x119
Fax: (905) 474 9223
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Website: www.sangoma.com
 

-Original Message-
From: Stig Even Larsen [mailto:[EMAIL PROTECTED]
Sent: November 25, 2005 3:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sangoma problems!?

I'm having problems connecting my Sangoma cards to our PRI (E1) interface.
It seems that the card get connected (green led), but Asterisk reports:
Status: Provisioned, Down, Active.

When I install another Sangoma card on the same system (pri_net), and
connect the two cards with a PRI cross-over cable both cards get connected
(green led) and Asterisk reports both spans:
Status: Provisioned, UP, Active

What could be wrong?
I'm I missing something?


zaptel.conf
;Telco PRI
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

;PBX
span=2,0,0,ccs,hdb3,crc4,yellow
bchan=32-46,48-62
dchan=47


zapata.conf (minimal)
[channels]

context = from-pstn
switchtype = euroisdn
group=0
signalling = pri_cpe
channel = 1-15
channel = 17-31

context = alcatel
switchtype = euroisdn
group=1
signalling = pri_net
channel = 32-46
channel = 48-62

Best regards,
Stig Even Larsen






David Yat Sin
Sangoma Technologies
(905) 474-1990 x119
(800) 388-2475 x119
Fax: (905) 474 9223
MSN: [EMAIL PROTECTED]
Email: [EMAIL PROTECTED]
Website: www.sangoma.com



-Original Message-
From: Stig Even Larsen [mailto:[EMAIL PROTECTED] 
Sent: November 25, 2005 3:21 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sangoma problems!?

I'm having problems connecting my Sangoma cards to our PRI (E1) 
interface. It seems that the card get connected (green led), but 
Asterisk reports:
Status: Provisioned, Down, Active.

When I install another Sangoma card on the same system (pri_net), and 
connect the two cards with a PRI cross-over cable both cards get 
connected (green led) and Asterisk reports both spans:
Status: Provisioned, UP, Active

What could be wrong?
I'm I missing something?


zaptel.conf
;Telco PRI
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16

;PBX
span=2,0,0,ccs,hdb3,crc4,yellow
bchan=32-46,48-62
dchan=47


zapata.conf (minimal)
[channels]

context = from-pstn
switchtype = euroisdn
group=0
signalling = pri_cpe
channel = 1-15
channel = 17-31

context = alcatel
switchtype = euroisdn
group=1
signalling = pri_net
channel = 32-46
channel = 48-62

Best regards,
Stig Even Larsen





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[Asterisk-Users] New mailing list: AstCallCenters

2005-11-28 Thread Lenz


Hello list,
this is just an announce of a new mailing list dedicated to deploying,  
running and managing real-
world Asterisk-based call centers. The mailing list is in English and  
allows knowledge sharing
for this very important - and yet somehow less considered - Asterisk  
deployment area.

The homepage is located at http://groups.yahoo.com/group/astcallcenters/
Yours,
l.


--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it

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Re: [Asterisk-Users] FAX difference IAXModem / Hylafax and spandsp app_rxfax

2005-11-28 Thread James Armstrong
Found out why there is no original DID set. It looks like while waiting 
for the incoming digits timeout (DID), we are getting the fax tone 
detect and it is sending a digit 'f' which immediately starts the fax 
extension before the incoming DID has been saved.


Is there a way to set in the zap config how many digits we are receiving 
so there is not timeout waiting for the last digit? We only get 7 digits 
coming in.


- James


James Armstrong wrote:
I looked into why I can't get the original DID number called when a fax 
is detected (so I can later route to the correct email address). There 
is a variable called FAXEXTEN that is created when a fax is detected, 
but it is not being populated with the original extension / did number 
called. It always has 's' as the original extension. Does anyone know 
how to fix this? I just want to use the zap fax detection so each person 
can have their personal DID number accept faxes and calls, then route 
the fax to their email address based on the DID number called.


- James


James Armstrong wrote:


In this example faxdetect is overwriting the DID.

So the trick then would be to somewhere early-on in your dialplan 
grab the DID into some variable, and then restore it after the fax 
detection occurs...


[default]
exten = _X.,1, SetVar(ORIGEXTEN=${EXTEN})
exten = s,2,Wait(3)
.
exten = fax,1,Dial(IAX2/ttyIAX0/${ORIGEXTEN})

Lee.




I am having no luck here. It seems the fax detection is overriding 
everything. I added the above and it never gets called. I guess it 
might be time to look at the Asterisk code and see if I can create 
another variable before the redirect happens.


-- Starting simple switch on 'Zap/1-1'
-- Redirecting Zap/1-1 to fax extension
-- Executing Dial(Zap/1-1, IAX2/999/|20) in new stack


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[Asterisk-Users] Realtime Extensions Problem

2005-11-28 Thread Dan Journo
Hi Guys,

Having a little problem with Realtime Extensions.

I've created the table, (using the same database as i use for realtime peers/users), however when a call comes through, the CLI shows the following warning:-

Nov 28 15:13:08 WARNING[7522]: res_config_mysql.c:623 mysql_reconnect: MySQL RealTime: Unable to select database: extensions. Still Connected.
and the call drops.

Has anyone seen this before, or suggest a solution?

Thanks
Dan
www.TextOver.com

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[Asterisk-Users] AGI script always returning 0

2005-11-28 Thread Benoît Mérouze

Hello,

I've noticed my AGI, in Perl, was always returning 0 even if exit from 
it with something else than 0.
On http://www.voip-info.org/wiki/view/Asterisk+cmd+AGI, it's said : 
[AGI] Returns -1 on hangup or if application requested hangup, or 0 on 
non-hangup exit. But I tried also to hang up during the execution of 
the AGI and it's the same.


I also tried to use setcallback like that:
-
# send callback reference
$AGI-setcallback(\callback);

# our callback function
sub callback(){
   my ($returncode) = @_;
   warn The call has ended ($returncode)\n;

   exit($returncode);
}
-
The $returncode is set to -1 when I hung up my phone, but in Asterisk 
logs there is still AGI Script pps.agi completed, returning 0.


Is that the normal behavior of Asterisk? Or is it possible to get -1 or 
any other code when finishing the execution of an AGI?


Thanks,
Benoit

--
Benoit Merouze
Network Software Developer
[EMAIL PROTECTED]

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Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian

2005-11-28 Thread Geo

It should build wcfxo. Not trying anything special.
I just follow the procedure !

When I reboot I have:

ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded 
!!!
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 2.46.2.5
HiSax: Layer2 Revision 2.30.2.4
HiSax: TeiMgr Revision 2.20.2.3
HiSax: Layer3 Revision 2.22.2.3


I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or 
whatever 
Is ISDN susbsystem needed for using fxo devices using fxs signalling with 
Asterisk ?


than
Zapata Telephony Interface Registered on major 196
wcfxo: disagrees about version of symbol zt_receive
wcfxo: Unknown symbol zt_receive
wcfxo: disagrees about version of symbol zt_ec_chunk
wcfxo: Unknown symbol zt_ec_chunk
wcfxo: disagrees about version of symbol zt_transmit
..
Testing 

modprobe zaptel = OK zaptel driver

but not wcfxo

and
ztcfg  -vvv

Zaptel Configuration
==

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

Yet my config is OK.
=
Installind Asterisk

make install
compiling OK but errors on zap
..
chan_zap.c:8935: error: dereferencing pointer to incomplete type
chan_zap.c:8936: error: dereferencing pointer to incomplete type
chan_zap.c:8950: error: dereferencing pointer to incomplete type
..





On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote:
 
 Well, thanks, it might be great your package yet I would like to know how to 
 adapt.
 I wouldn't like to rewrite Debian neither Asterisk but is somebody able to 
 advice 
 how you define modules in zconfig.h or whatever ?
 Any tip ?
 Geo

Why would you need to define modules? The package builds wcfxo. 
What exactly do you try to do?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian

2005-11-28 Thread Geo

It should build wcfxo. Not trying anything special.
I just follow the procedure !

When I reboot I have:

ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded 
!!!
HiSax: Linux Driver for passive ISDN cards
HiSax: Version 3.5 (module)
HiSax: Layer1 Revision 2.46.2.5
HiSax: Layer2 Revision 2.30.2.4
HiSax: TeiMgr Revision 2.20.2.3
HiSax: Layer3 Revision 2.22.2.3


I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed or 
whatever 
Is ISDN susbsystem needed for using fxo devices using fxs signalling with 
Asterisk ?


than
Zapata Telephony Interface Registered on major 196
wcfxo: disagrees about version of symbol zt_receive
wcfxo: Unknown symbol zt_receive
wcfxo: disagrees about version of symbol zt_ec_chunk
wcfxo: Unknown symbol zt_ec_chunk
wcfxo: disagrees about version of symbol zt_transmit
..
Testing 

modprobe zaptel = OK zaptel driver

but not wcfxo

and
ztcfg  -vvv

Zaptel Configuration
==

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)

1 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

Yet my config is OK.
=
Installind Asterisk

make install
compiling OK but errors on zap
..
chan_zap.c:8935: error: dereferencing pointer to incomplete type
chan_zap.c:8936: error: dereferencing pointer to incomplete type
chan_zap.c:8950: error: dereferencing pointer to incomplete type
..





On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote:
 
 Well, thanks, it might be great your package yet I would like to know how to 
 adapt.
 I wouldn't like to rewrite Debian neither Asterisk but is somebody able to 
 advice 
 how you define modules in zconfig.h or whatever ?
 Any tip ?
 Geo

Why would you need to define modules? The package builds wcfxo. 
What exactly do you try to do?

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Download Ringtones for 7960's?

2005-11-28 Thread Frank McCarthy
Does anybody know where I can download ringtones for Cisco 7960's? Need
to be .pcm files.
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[Asterisk-Users] Can 'spandsp' ack as an intermediary between a fax machine and a TDM400P?

2005-11-28 Thread Chuck Bunn

Hi,

I understand that a fax machine cannot connect through a Digium TDM400p 
card (FXS connected to fax and FXO connected to a  pots line) but can 
spandsp send and receive faxes as an intermediary between the pots line 
and the fax machine.


Thanks
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Re: [Asterisk-Users] sound problem, please help!

2005-11-28 Thread Rusty Dekema
Hi,

I have noticed that most mobile phones (GSM and CDMA at least) seem to
have a tendency to interrupt the incoming audio stream when the
microphone levels get louder than a certain threshold (such as when you
are speaking into it). I do not know exactly why this happens, nor
whether it is something that happens in the handset (the handset mutes
the incoming audio, which doesn't make much sense) or in the network
(maybe to try to save bandwidth?). 
Also, the GSM codec is intended to intelligibly encode human speech,
not general audio signals such as music. GSM uses approximately 6-8
times less bandwidth than G.711u/a, and the degradation in the quality
of music and other non-speech signals is one of the tradeoffs that we
make in order to not have the cost of mobile calls increase 6-8 times
over.

Maybe if Europe was as backward with its mobile systems as the United
States currently is, you could try to get your users to call your
system from analog mobile phones... They don't use digital compression
(by definition, I suppose) and would probably work much better for your
specific application ;).

-Rusty


On 11/25/05, Esteban Maestre [EMAIL PROTECTED] wrote:
Hi all!I have a strange problem when using asterisk. I have configured asteriskto receive calls (FX0). In my configuration, I want asterisk to play musicwhileI record the caller's speech. If the caller does the call from a
fixed line telephone, there is no problem, but in case the caller does thecall from a mobile GSM phone, the quality of the music he hears becomes sobad, and even more when he speaks.I have tried several codecs.
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[Asterisk-Users] Trunk SIP howto ?

2005-11-28 Thread Noc Phibee

Hi

anyone know if a Trunk SIP howto are created ?

I have 8 VoIP account with for all 1 login/pass per number.
i want add into my asterisk but not know where ;=)

Other questions:

my supplierhave a dns:sip.phonesystems.net
this name have 2 IP address
it's not a problems for Asterisk that he have registred on
the first ip address and receive information of the second ?

Thanks for your help


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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Noc Phibee

Thanks sergio for your answer.

But cisco france say me that i cant' bye SmartNet contract on this product.
Only one solution are possible: Bye a special contract at $180.00 ...
Pff i can bye a new equipment with this price hihihi

i can't guest the latest firmware, for me i thinks that the solution are buy
new voip phone and put the 7910 in Dead

If anyone know a solution for get the latest firmware, mail me

Bye





Sergio Chersovani a écrit :

Noc Phibee ha scritto:


it's possible to upgrade the firmware of a cisco 7910 with asterisk ?


You need the legal firmware upgrade file
download the chan_sccp code from http://chan-sccp.berlios.de
configure it and use the imageversion param to upgradde the phone 
firmware.


Of course you need a tftpserver and if you run a tftpserver you just 
need a SEPmac to upgrade the phone

So the correct answer is:
you don't need a CCM nor asterisk to upgrade a cisco phone firmware.
You just need the firmware file, a tftpserver and a configuration file 
(SEPmac)


take a look here
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx


Sergio
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[Asterisk-Users] ip phones

2005-11-28 Thread John Fraser
Hi all,

 Does anybody have any info on a decent quality sip hard phone that is 
headset compatible?

 Thank you
 John
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[Asterisk-Users] How does DTMF get sent over a PRI in Asterisk

2005-11-28 Thread James Sizemore
I am trying to trouble shoot some problems with DTMF over PRI. I have a 
digium wct1xxp card and these lines in extensions.conf:


exten = 5556000,1,Record(testtone:gsm)
exten = 5556000,2,Wait(2)
exten = 5556000,3,Playback(testtone)

I called in over the PSTN --to--  Asterisk. I did a pri debug, from 
asterisk 1.2.0 console and a few debug lines came up when the call 
connects (but not all the q931 message that should have been there). But 
nothing came up when hitting the dtmf keys. Also when I listen to the 
play back the recorded tones are little more then chirps not long enough 
for a human ear to distinguish tone.


My two question are how does DTMF get send over a PRI (inband? q931 
message?) and how would I go about seeing the duration that was sent to 
me so that I know weather the problem is Asterisk or my telcos!

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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Tony Hoyle

Noc Phibee wrote:

Thanks sergio for your answer.

But cisco france say me that i cant' bye SmartNet contract on this product.


You can, but only in the US I believe.  I've never found any deal less 
than £150 (UK).



Only one solution are possible: Bye a special contract at $180.00 ...
Pff i can bye a new equipment with this price hihihi


It's not so bad... you do get access to firmware to all cisco devices 
with that, so if you have more than one device it becomes worth it.


You also get access to cisco TAC.  Their support sucks in my experience, 
but it's far better than having no support at all when your hardware 
fails (cisco won't even give a manufacturers guarantee unless you have a 
smartnet contract).


Tony
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Sergio Chersovani

Noc Phibee ha scritto:

But cisco france say me that i cant' bye SmartNet contract on this 
product.


why not?
You can buy those smartnet contract via internet. You just need to mail 
US cisco and ask them for the contract activation.



Only one solution are possible: Bye a special contract at $180.00 ...


buy CON-SW-VPKG1 59 euros in europe


Pff i can bye a new equipment with this price hihihi


yep that is cisco

i can't guest the latest firmware, for me i thinks that the solution 
are buy

new voip phone and put the 7910 in Dead


Yes you are right

Sergio
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Re: [Asterisk-Users] sound problem, please help!

2005-11-28 Thread Esteban Maestre
Hello, Rusty!

Thanks for your reply... You have been the only one ;)

Hahaha, it could become dangerous... ;)

Well, I have been investigating it a little bit, and I guess the main
reason is something related to what you have pointed in the first
paragraph.
I've found some interesting information here:

http://www.privateline.com/PCS/GSM08.html

There is an interesting article I've already tried to post, but toy can
find it easily in the www above.

regards,
-esteban-


 I don't think you'd want to hear most of what I say when I am on hold,
 especially if I am on hold with a telephone company! It tends to be
 somewhat
 on the profane side. *grin*

 -Rusty

 On 11/25/05, Esteban Maestre [EMAIL PROTECTED] wrote:

 kind of... ;)
 I want to know what the people say when they are waiting... :P

 do you have any idea on what the problem could be?

 -esteban-


   Original Message 
  From: Esteban Maestre [EMAIL PROTECTED]
  To: asterisk-users@lists.digium.com
  Sent: Friday, November 25, 2005 11:22 AM
  Subject: [Asterisk-Users] sound problem, please help!
 
  Hi all!
 
  I have a strange problem when using asterisk. I have configured
  asterisk to receive calls (FX0). In my configuration, I want asterisk
  to play music while  I record the caller's speech.
 
  Dialup-karaoke? :-)
 
  Leif
 
 
 
 


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RE: [Asterisk-Users] ip phones

2005-11-28 Thread Anders Svensson
The only one I can think of to decent price level is the Grandstream GXP
2000. Also have headset jack¨

Anders

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: den 28 november 2005 17:27
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ip phones

Hi all,

 Does anybody have any info on a decent quality sip hard phone that is 
headset compatible?

 Thank you
 John
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Noc Phibee

Thanks all for your answer ...

all smartnet contrat have access to all firmware in voip ?

thanks



Ryan Amos a écrit :
Cisco phones are not ideal for single-phone setups. If you were to have a lot of them, a $180 support contract is no big deal... However, for Europeans, there should be an $8 online-only support contract that gives you access to file downloads only. Being an American, 


This should be enough, however if you are only wanting a small number of phones 
you might want to look elsewhere. The main advantage of Cisco's phones comes 
when installing a large number of them, as the central management is ideal in 
an office PBX environment.

Try this part number though: CON-SNT-PKG1-VS Supposedly costs 66 euros from 
wstore.fr (I found this in an old e-mail asking about smartnet contracts on the 
chan_sccp mailing lists.) Best of luck!

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Noc Phibee
Sent: Monday, November 28, 2005 10:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

Thanks sergio for your answer.

But cisco france say me that i cant' bye SmartNet contract on this product.
Only one solution are possible: Bye a special contract at $180.00 ...
Pff i can bye a new equipment with this price hihihi

i can't guest the latest firmware, for me i thinks that the solution are buy
new voip phone and put the 7910 in Dead

If anyone know a solution for get the latest firmware, mail me

Bye


  


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[Asterisk-Users] IAX jitterbuffer and trunking settings between 1.0.9 and 1.2

2005-11-28 Thread Chris Bagnall
Hello all,

Since upgrading a couple of our servers to 1.2 I've noticed problems when
talking to users on 1.0.9 servers. The servers are connected via IAX2 with
trunking and jitter buffer enabled (jitter buffer on default settings).

Reading through posts in the list archives, there are a number referring to
problems with 1.0.9 and the jitter buffer when used in conjunction with
trunking.

Are there any recommended settings to use for the following scenarios?
1.0.9 - 1.0.9
1.2.0 - 1.0.9
1.2.0 - 1.2.0

Thanks in advance.

Regards,

Chris
-- 
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This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Download Ringtones for 7960's?

2005-11-28 Thread Bob Goddard
On Monday 28 Nov 2005 15:39, Frank McCarthy wrote:
 Does anybody know where I can download ringtones for Cisco 7960's? Need
 to be .pcm files.

Download the ringtone generator from Grandstream and make your own.



B
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Re: [Asterisk-Users] ip phones

2005-11-28 Thread Michael Graves
On Mon, 28 Nov 2005 08:26:57 -0800, John Fraser wrote:

Hi all,

 Does anybody have any info on a decent quality sip hard phone that is 
headset compatible?

 Thank you
 John

Aastra 480i ( Ilove it!), Polycom IPx00/x01 series, Snom's all provide
for headsets. Most via RJ style connections, some (SNOM) both RJ and
2.5mm or 3.5mm jacks.

Michael

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

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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[Asterisk-Users] how to stop ringing while talking

2005-11-28 Thread erkan kolemen
hi,while a user talking on, if he gets a new call from queue he hears some  noises about this call. I think this happens to inform user about new  coming call. But it boring user... So can i stop this noise?Thanks.-ek  
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RE: [Asterisk-Users] Does it mean I was blocked by STUN?

2005-11-28 Thread Juan Janczuk
Hi.
I think the best way to do it, is just a IAX2 between the 2 *'s.

Regards.
Juan.

 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] nombre de Hiu Yen Onn
 Enviado el: Domingo, 27 de Noviembre de 2005 10:50 p.m.
 Para: asterisk-users@lists.digium.com
 Asunto: [Asterisk-Users] Does it mean I was blocked by STUN?


 Hi all,

 I have 2 respectively networks, LAN A and LAN B, connected via my
 wireless links and routers. I have setup an asterisk machine at LAN A.
 It works fine when i was in LAN A. But, when i was in LAN B, xlite
 client can get connected to the server. But, it has no sound when i try
 to make an echo test. Does it mean, i was blocked by STUN? I have a wild
 search on google, i found that asterisk doesnt really support STUN. What
 is the workaround to make two network clients enjoy the intercalling via
 asterisk? IAX please advise... thanks
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RE: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Chris Bagnall
 You can, but only in the US I believe.  I've never found any 
 deal less than £150 (UK).

I was quoted £36 a couple of weeks ago by one of the Cisco resellers a
google search provided me with, if that's any help. I can't remember the
company name I'm afraid...

Regards,

Chris
-- 
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This email is made from 100% recycled electrons


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[Asterisk-Users] Call progress from sip gsm gateway to pri interface - doesn't get through

2005-11-28 Thread Robert Rozman

Hi,

we have following setup : PBX - Parlay -ISDN PRI- Asterisk -SIP-  GSM 
Gateway


Call comes from PBX through Parlay to Asterisk and it routes it over SIP to 
GSM gateway. GSM gateway gives back call progress (it takes some time to 
ring or get through), but this info won't get back to Parlay on ISDN PRI 
interface (Digium PRI card), so Parlay after some timeout disconnects 
call


We guess that this setup should work, but we're not sure. Anyone with 
working setup like this? Anyone with experience of call progress getting 
from SIP to PRI or BRI interfaces ?  Any advice or pointer to more info ?


Thanks in advance,

regards,

Rob.

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RE: [Asterisk-Users] ip phones

2005-11-28 Thread Kerry Garrison
Depends on the type of headset. The Grandstream GXP-2000 and Liksys SPA-941
have headphone jacks on them, most phones are compatible with Plantronics
univeral headsets.

Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: Monday, November 28, 2005 8:27 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ip phones

Hi all,

 Does anybody have any info on a decent quality sip hard phone that is
headset compatible?

 Thank you
 John
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Bob Goddard
On Monday 28 Nov 2005 16:29, Tony Hoyle wrote:
 Noc Phibee wrote:
  Thanks sergio for your answer.
 
  But cisco france say me that i cant' bye SmartNet contract on this
  product.

 You can, but only in the US I believe.  I've never found any deal less
 than £150 (UK).

I guess that should depend as to whether it is hardware or software only.

  Only one solution are possible: Bye a special contract at $180.00 ...
  Pff i can bye a new equipment with this price hihihi

 It's not so bad... you do get access to firmware to all cisco devices
 with that, so if you have more than one device it becomes worth it.

And it is also illegal.


B
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Re: [Asterisk-Users] Problem with ADIT 600 and FXO configuration

2005-11-28 Thread C F
What does the TE406 leds indicate?

On 11/28/05, William K. Volkman [EMAIL PROTECTED] wrote:
 I've looked through the archives of the mailing list for the
 last year and although informative I've not been successful
 at get this to work.  We had a working Asterisk PBX system
 with 3 Digium X101P FXO lines and two TDM400P FXS cards.
 I've setup an ADIT 600 with an 8 port FXO card (and an
 8 port FXS card not currently installed).  We are going
 to be adding a T1 for incoming calls this week. I removed
 two of the X101P cards and installed a TE406P.  I'm using
 Asterisk 1.0.9 (and matching zaptel, libpri) from tar files.

 /etc/zaptel.conf has this configuration:
 span=1,1,0,esf,b8zs,yellow
 span=2,0,0,esf,b8zs
 span=3,0,0,esf,b8zs
 span=4,0,0,esf,b8zs
 #Modular unit, first card is FXO
 fxsks=1-3
 unused=4-8
 #Modular unit, 1 FXS cards
 unused=9-16
 unused=17-24
 unused=25-48,49-72,73-96
 fxsks=97
 fxoks=98-101
 fxoks=102-105

 /etc/asterisk/zapata.conf has this:
 group = 0
 signalling=fxs_ks
 context = incoming
 busydetect = yes
 overlapdial = no
 channel = 1-3

 signalling=fxs_ks
 channel = 97  ;X100P

 group = 1
 signalling = fxo_ks
 context = internal
 ;TDM400P
 callerid = Available 200
 channel = 98-100
 callerid = x
 channel = 101
 ;TDM400P
 callerid = x
 channel = 102
 callerid = x
 channel = 103

 Parts of my adit configuration:
 -Setting slot a.

 set a:1 up
 set a:1 fdl none
 set a:1 lbo 4
 set a:1 framing esf
 set a:1 id Inbound
 set a:1 linecode b8zs
 set a:1 loopdetect csu
 set a:1:1-24 side drop
 set a:1:1-24 type voice
 set a:1:1-24 signal ls
 set a:2 up
 set a:2 fdl none
 set a:2 lbo 1
 set a:2 framing esf
 set a:2 id Outbound PBX
 set a:2 linecode b8zs
 set a:2 loopdetect csu
 set a:2:1-24 side drop
 set a:2:1-24 type voice
 set a:2:1-24 signal ls

 -Setting slot 1.

 set 1:1-8 signal lscpd
 set 1:1-8 txgain -3
 set 1:1-8 rxgain -6

 -Setting primary and secondary clock sources.

 set clock1 a:1
 set clock2 internal

 -Setting the system idle pattern for DS0s.

 set idle 0xff

 -Making connections.

 connect a:2:1-3 1:1-3

 Inbound calls just ring and ring (the leds on the ADIT change
 state) however asterisk doesn't respond.  Attempts to make
 outgoing calls get:
 -- Executing Dial(SIP/202-ba07, Zap/g0/5551212) in new stack
 Nov 28 02:54:45 NOTICE[8627]: app_dial.c:764 dial_exec: Unable to create
 channel of type 'Zap'
   == Everyone is busy/congested at this time
 -- Executing Congestion(SIP/202-ba07, ) in new stack
   == Spawn extension (from-sip, 95942060, 3) exited non-zero on
 'SIP/202-ba07'
 -- Executing Hangup(SIP/202-ba07, ) in new stack

 I've tried just about all combinations of gs/ls/ks for the
 signalling to no avail.  Here is the output of status:

  status a:2:1-3

 DS0 Rx AB  Tx AB  Signal  T1 TP
 --- -  -  --  -  --
 a:2:1 01 01  LS   TrafficN
 a:2:2 01 01  LS   TrafficN
 a:2:3 01 01  LS   TrafficN

  status 1:1-3

 FXORx AB  Tx AB  Signal=T1 Sig  T1 TP
 ----  -  --  -  --
 1:1  01 01   LSCPD = LS Traffic N
 1:2  01 01   LSCPD = LS Traffic N
 1:3  01 01   LSCPD = LS Traffic N

  show connect a:2:1-3
 FromDescDescTo
  ---  --  -  -
   A:02:01LS VOICE   DS0 -- FXOVOICE LSCPD   1:01
   A:02:02LS VOICE   DS0 -- FXOVOICE LSCPD   1:02
   A:02:03LS VOICE   DS0 -- FXOVOICE LSCPD   1:03

 Can anyone spot what I've got wrong?  Any suggestions or hints
 welcome.

 Thanks,
 William.


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[Asterisk-Users] Emailed voicemail messages not being deleted

2005-11-28 Thread Dustin Wenz
According to the Asterisk wiki, adding the delete=yes option to a  
voicemail definition should automatically delete messages after they  
are emailed. This is the format that I'm using:

101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes

When I leave a message at mailbox 101, it gets emailed correctly but  
is not removed from the voicemail box. Am I missing something here,  
or is this feature broken?


- .Dustin Wenz
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[Asterisk-Users] PROGRESS with cause code 31 received

2005-11-28 Thread Dana Olson
I have been trying to work this problem out with my IAX provider.

I dial a toll-free number, ex: 1-888-876-6262, and I get a due to technical difficulties message.

I set my debug level to 9, and all I see when I dial out is this:

-- Executing Dial(SIP/27-51de, IAX2/voctel/1766262||T) in new stack-- Called voctel/1766262-- Call accepted by 204.14.18.189
 (format ulaw)-- Format for call is ulaw-- IAX2/voctel-3 is proceeding passing it to SIP/27-51de-- IAX2/voctel-3 is making progress passing it to SIP/27-51de-- Hungup 'IAX2/voctel-3'  == Spawn extension (longdistance, 1766262, 1) exited non-zero on 'SIP/27-51de'


What my IAX provider sees on the other end is this:

-- Executing Dial(IAX2/tor-hub-13, Zap/G1/1766262||g) in new stack 
-- Requested transfer capability: 0x00 - SPEECH 
-- Called G1/1766262 
-- Zap/21-1 is proceeding passing it to IAX2/tor-hub-13 
-- PROGRESS with cause code 31 received 
-- Zap/21-1 is making progress passing it to IAX2/tor-hub-13 
-- Hungup 'Zap/21-1' 

I did a search through the mailing list and in the wiki. I found that cause code is used to report a normal event only when no other cause in thenormal class applies. and 
#defineAST_CAUSE_NORMAL_UNSPECIFIED			31.

I am running Asterisk 1.2.0 and I am not sure what my provider is using, some version of HEAD is all I know.

I am at a loss... I don't know the last time I tried to dial a toll-free from here, but it was working.

Can anyone help steer me in the right direction?

Thanks!

Dana

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[Asterisk-Users] SNOM Phones MWI, Hold Retrieve buttons not working with Asterisk v1.2

2005-11-28 Thread Sascha Deri
We just upgraded to Asterisk 1.2 a few days ago. And now the Retrieve 
voicemail and hold buttons on our SNOM 360 phones are no longer 
working.  When you put a caller from one of our zaptel lines on hold it 
hangs up on them immediately. Interestingly, if you put an internal 
extension on hold it does not drop the call.


Additionally, the Retrieve voicemail butotn on the phones no longer 
work.  The MWI (Message Waiting Indicator) lights up, but when you press 
the button you get Not Found sip:asterisk@ and busy signal.


Any ideas on what might be wrong on how to fix it?

Thanks!
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[Asterisk-Users] Re: Problem connecting Two * servers with SIP (used to be: SIP Forward)

2005-11-28 Thread Waldo Rubinstein

I apologize for the resend. I haven't received much feedback from this.

I also noticed that what I'm getting is the caller id as the caller  
name and the sip peer name as the caller id number.


Does anyone have any ideas/suggestions?

Thanks,
Waldo

On Nov 26, 2005, at 2:52 AM, Waldo Rubinstein wrote:


Hi guys,

I'm trying to forward a call from one * server to another using SIP.

Everything works when I use fromuser in the sip entry of the *  
forwarding the call. The problem is that when the receiving * sends  
the call to the UA, it puts the caller to be the value of fromuser  
instead of the caller-id (as documented).


If I remove the fromuser, then the calls are denied because of  
wrong password on authentication for INVITE to  
'{calleridnum} ...'. Meaning, it's trying to authenticate the call  
based on caller-id instead of the peer name.


How can I set it up to authenticate without necessarily using  
fromuser?


Thanks,
Waldo



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[Asterisk-Users] Asterisk project converts to Subversion version control system

2005-11-28 Thread Asterisk Development Team
The Asterisk development team is pleased to announce that we have 
migrated our project repositories and development processes over to the 
Subversion version control system!


Effective immediately, the primary source code distribution point for 
Asterisk, Zaptel and other related projects (other than release tarballs 
of course) will be http://svn.digium.com.


The actual SVN repositories are available at http://svn.digium.com/svn, 
and there is a ViewCVS web viewer available at 
http://svn.digium.com/view. There is a separate repository for each 
major project, and each repository is organized in the typical 
Subversion fashion... for example, the Asterisk repository is organized 
as follows:


http://svn.digium.com/svn/asterisk/trunk (was CVS HEAD)
http://svn.digium.com/svn/asterisk/branches/1.2 (was CVS v1-2)
http://svn.digium.com/svn/asterisk/tags/1.2.0 (was CVS v1-2-0)

Other branches and tags are named similarly. Builds of Asterisk made 
from the new repositories will report a 'show version' tag made of the 
SVN branch name and the repository revision number that was checked out 
(unlike the CVS 'show version' tags which incorporated the date/time of 
checkout).


The 'asterisk-cvs' mailing list has been renamed to 'svn-commits' and 
will continue to receive commit messages for the all the major projects 
on our SVN server (existing subscriptions are still in effect). In 
addition, there are new project specific commit mailing lists as well:


  asterisk-commits
  asterisk-addons-commits
  zaptel-commits
  libpri-commits
  libiax2-commits

All of these lists are available on lists.digium.com. Additionally, the 
commit messages will contain 'X-SVN-Author' and 'X-SVN-Branch' mail 
headers to allow you to sort/filter the commit messages in any way you wish.


One of the major benefits of this transition is that we will be opening 
up 'developer branches' for Asterisk Development Team members to be able 
to work on projects and make them available for public review, testing 
and participation; look for another announcement later this week when 
that process is ready.


For the near future, we will continue to provide access to source code 
via CVS using the same servers/paths that you have previously been 
using; once every day, the relevant Subversion branches will be copied 
over into CVS and brought up to date. We expect to keep updating CVS 
HEAD this way for three to six months; the other branches will be 
maintained for six to nine months. However the CVS repositories will be 
updated in a single commit each day and will not contain any detailed 
revision history for the changes that are made. We encourage all users 
to transition to using Subversion for tracking development as soon as 
possible.


(Special thanks to chipig, sussman, darix, jerenkrantz, eh, mbk and the 
others on #svn-dev who helped solve some sticky issues on Saturday 
evening of Thanksgiving weekend G)

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[Asterisk-Users] Problem with pulses dialing on asterisk 1.2

2005-11-28 Thread Cyrille DERORY
I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP softphones, 
7905G cisco SCCP and analog phone( DTMF dialing). All is working nice, 
however when I change DTMF for an analog pulse dialing,my analog phone 
is not working.


I've found the following :

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
pulse=yes

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing
pulsedial=yes

What is correct : pulse=yes or pulsedial=yes ?

However none are working !

I probably miss something.
I need your help.

Is there any french users with pulses dialing on analog phones ?

Thanks.
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Re: [Asterisk-Users] Pros and Cons of T1/E1 cards

2005-11-28 Thread Joe Pukepail
I haven't heard of this product before so I did some searches on the Internet, this card is $5,400 for a single span T1 card? ouch!
http://www.eiconworks.com/DivaServerV-PRI_T1%20.asp
On 11/25/05, David Waugh [EMAIL PROTECTED] wrote:
Hi John,I'm going to have to disagree with some previous posts.The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the load of the call handling, echo cancellation etc as this is all processed on board on the card, and not on the central CPU of the computer.
You can use the CAPI interface of the card combined with chan_capi_cm with the card.I have not found any problems when using different kernels or different versions of asterisk.I have one setup in our test lab here at Eicon with Asterisk so it does work!
You can have up to 8 Diva Server cards in once machine - including a mixture of the analog and BRI cards.The Diva Server cards in two variants - the V-Series if you only want to use them with Voice based applications and the normal All-in-one cards if you want to do fax and RAS too.
If you need any more information let me know, and I will assist furtherDavid-Original Message-From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of JohnDaragonSent: 25 November 2005 00:46To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Pros and Cons of T1/E1 cardsHi;We're looking to standardise on a single family of E1 PRI cards.I guess our options are :Digium / Zaptel / libpriSangoma/ Zaptel / Wanpipe
AVM/ CAPIeIcon/ CAPIJunghanns/ BristuffCan anyone share any comparative experience of these, please ? Do theydiffer much in terms of interrupt requirement, CPU load c ?
Any info gratefully received.jd--John Daragon[EMAIL PROTECTED]argv[0] limitedLambs Lawn Cottage,Staple Fitzpaine,Taunton,TA3 5SL,UK
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[Asterisk-Users] AGI + CDR

2005-11-28 Thread Innocent Evil
Hi,

I have an AGI script that is called after receving a call on a channel.
And my script executel AGI cmd Dial to make another call.
Is there any reason not to have CDR record for the call that was initiated in 
the AGI script?
Or I am just missing something basics .

Thanks,


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Re: [Asterisk-Users] Problem with pulses dialing on asterisk 1.2

2005-11-28 Thread John Novack



Cyrille DERORY wrote:

I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP 
softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All is 
working nice, however when I change DTMF for an analog pulse 
dialing,my analog phone is not working.


I've found the following :

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
pulse=yes

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse+dialing 


pulsedial=yes


Pulsedial=yes works

You should be aware that inserting a w into the dialstring does NOT 
work with pulsedial, however, so if your PSTN connection is a little 
slow, you may get misdials


Remember you need to restart after making the change.
To be really safe, reboot.

John Novack


What is correct : pulse=yes or pulsedial=yes ?

However none are working !

I probably miss something.
I need your help.

Is there any french users with pulses dialing on analog phones ?

Thanks.
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Re: Re: Re: [Asterisk-Users] Zaptel errors on Debian

2005-11-28 Thread Tzafrir Cohen
Hi

On Mon, Nov 28, 2005 at 04:35:34PM -0800, Geo wrote:
 
 It should build wcfxo. Not trying anything special.
 I just follow the procedure !
 
 When I reboot I have:
 
 ISDN subsystem Rev: 1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded 
 !!!
 HiSax: Linux Driver for passive ISDN cards
 HiSax: Version 3.5 (module)
 HiSax: Layer1 Revision 2.46.2.5
 HiSax: Layer2 Revision 2.30.2.4
 HiSax: TeiMgr Revision 2.20.2.3
 HiSax: Layer3 Revision 2.22.2.3
 
 
 I don't need ISDN, I do not configure any ISDN and I have no ISDN BRIstuffed 
 or whatever 
 Is ISDN susbsystem needed for using fxo devices using fxs signalling with 
 Asterisk ?
 

What's this ISDN driver doing here?

A look at lspci will show you:

 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface

But by now you already knowthat this line represents your X100P card
that hapens to have the same PCI ID as that TigerJet device. 'lspci -n'
will show that the actual device has vendor ID e159 and product ID 1 . 
'grep e159 /lib/modules/`uname -r`/modules.pcimap' will show that a
number of zaptel modules look for devices with those vendor/product IDs
but with some specific subvendor IDs and that the hisax driver tries to
load them all. 

hotplug uses that information (extracted from the modules at depmod
time) to load modules by bus IDs. Don't want it? blacklist it:

  echo hisax  /etc/hotplug/blacklist.d/local

Consider blacklisting other modules whose automatic modprobe seems
unnecessary/pointless in just the same way (or $EDITOR
/etc/hotplug/blacklist )

 
 than
 Zapata Telephony Interface Registered on major 196
 wcfxo: disagrees about version of symbol zt_receive
 wcfxo: Unknown symbol zt_receive
 wcfxo: disagrees about version of symbol zt_ec_chunk
 wcfxo: Unknown symbol zt_ec_chunk
 wcfxo: disagrees about version of symbol zt_transmit

This beats me: version mipatch between zaptel and wcfxo ?

One possible guess: you installed everything from one place. And then
you compiled it again (without wcfxo this time) and reinstalled.

Are you using m-a?

 ..
 Testing 
 
 modprobe zaptel = OK zaptel driver
 
 but not wcfxo
 
 and
 ztcfg  -vvv
 
 Zaptel Configuration
 ==
 
 Channel map:
 
 Channel 01: FXS Kewlstart (Default) (Slaves: 01)
 
 1 channels configured.
 
 ZT_CHANCONFIG failed on channel 1: No such device or address (6)

Seems that wcfxo is not loaded. 

  lsmod | grep zaptel

 
 Yet my config is OK.
 =
 Installind Asterisk
 
 make install
 compiling OK but errors on zap
 ..
 chan_zap.c:8935: error: dereferencing pointer to incomplete type
 chan_zap.c:8936: error: dereferencing pointer to incomplete type
 chan_zap.c:8950: error: dereferencing pointer to incomplete type
 ..
 
 
 
 
 
 On Sun, Nov 27, 2005 at 07:40:04PM -0800, Geo wrote:
  
  Well, thanks, it might be great your package yet I would like to know how 
  to adapt.
  I wouldn't like to rewrite Debian neither Asterisk but is somebody able to 
  advice 
  how you define modules in zconfig.h or whatever ?
  Any tip ?
  Geo
 
 Why would you need to define modules? The package builds wcfxo. 
 What exactly do you try to do?
 
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[Asterisk-Users] Help connecting Avaya S8700 and Asterisk through H.323 trunk

2005-11-28 Thread Pablo Chacón
Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk
(using channel oh323).
I can make calls from S8700 H323 extension to Asterisk SIP phone using
G711a codec but when I try to make a call from SIP phone to S8700
extension I listen one ringing tone and the call is dropped.
Can anybody help me???

I see some errors in log but I can't resolve it:

1. -- H.323 call 'ip$localhost/29416-70919cfe' cleared, reason 8
(Transport failure)

2.  0:23.635  H225 Caller:8264258 H225Connect of H245
failed: Connection refused

Full logs are attached below

Thanks.



---
oh323.conf:

;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
connectPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
;   rtp.conf
;
udpStart=1
udpEnd=2
;
; Enable fast start (yes,no).
;
fastStart=yes

;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=100
inboundMax=100
simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=10
libTraceLevel=10
libTraceFile=/var/log/asterisk/h323.log
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is
the zone name.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   gatekeeper's DNS name,
;   gatekeeper's ip,
;   GKID:gatekeeper's id
;   gatekeeper's id@gatekeeper's name or address
;
gatekeeper=DISABLE
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;   INBAND  -
;
userInputMode=INBAND
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=voip-h323

;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
;alias=asterisk
;
; Aliases/prefixes routed in all-aliases context.
;
; Aliases/prefixes routed in more-aliases context.
;
; Aliases/prefixes routed in all-prefixes context.
;
; Aliases/prefixes routed in more-stuff context.
;

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every codec option may have a frames option
; associated with it.
; Valid values for the codec option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G726-   G.726(32k)
;   G72616K -   G.726(16k)
;   G72624K -   G.726(24k)
;   G72632K -   G.726(32k)
;   

[Asterisk-Users] Wrong usage of [] in the extension?

2005-11-28 Thread Rusty Dekema
Hello,

I am trying to set up my dialplan in such a manner that calls to
numbers in the form 1-NPA-NXX- will only go through if the NPA
dialed is a geographical NPA in the continental United States. 

I have collected a list of all NPAs that I want to allow, and have made
the following dialplan entries (below). Calls that match the pattern do
go through, but for each digit that I dial, I get the following error
on the Asterisk console: 

WARNING[7045]: pbx.c:718 ast_extension_close: Wrong usage of [] in the extension

or

WARNING[7045]: pbx.c:699 ast_extension_match: Wrong usage of [] in the extension

If I comment out the three entries below, the errors stop (and of course the calls do not go through). 

I have checked that each dialplan entry is on one line in the file;
there are no accidental carriage returns or newlines present.

Does anyone know what I am doing wrong? I realize that I have a rather
large pattern defined, but it seems to be using correct syntax as far
as I can tell. If there is a better way to do this, or if someone sees
what I am doing incorrectly, could you please let me know? 

Thanks very much,
Rusty



Extensions.conf excerpt: 


exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,1,Playback(local/ding
ding)

exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Dial(IAX2/[EMAIL PROTECTED]
jet/${EXTEN},60)

exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Goto(call-dispositi
on|s|1)

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Re: [Asterisk-Users] Emailed voicemail messages not being deleted

2005-11-28 Thread Gonzalo Servat
On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote:
 According to the Asterisk wiki, adding the delete=yes option to a
 voicemail definition should automatically delete messages after they
 are emailed. This is the format that I'm using:
 101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes
[snip]

Try:

101 = ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes

(notice the extra comma after the email address)

I believe the setting that goes in between the empty commas is the
pager email address

Hope this helps.

Cheers,
Gonzalo
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[Asterisk-Users] cdr_manager.conf

2005-11-28 Thread Innocent Evil
Hello,

While I was trying to get right CDR record from AGI script, I came across 
cdr_manager.conf
I am trying to learn about cdr_manager.conf

What is the purpose of cdr_manager.conf?
How I can configure it?

I did google, really didn't have very good luck.

Would anybody please write couple of sentenses regarding this.
Any link on documentaion, tutorial would be great help.

Thanks,


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Re: [Asterisk-Users] Help connecting Avaya S8700 and Asterisk through H.323 trunk

2005-11-28 Thread BJ Weschke
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote:
 Hi I'm trying to connect Avaya S8700 and Asterisk through H323 trunk
 (using channel oh323).
 I can make calls from S8700 H323 extension to Asterisk SIP phone using
 G711a codec but when I try to make a call from SIP phone to S8700
 extension I listen one ringing tone and the call is dropped.
 Can anybody help me???


 I've had greater success increasing the number of frames in an RTP
packet when dealing with the med pro resources on the S8700.

 Also, make sure you're sending the call to the IP that is bound to
the CLAN board that also has the signaling group you're trying to call
into bound to it. With the connection refused here it seems like you
might be trying to send the call to the IP of the med pro board
instead of a CLAN board.

 BJ


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[Asterisk-Users] DTMF errors

2005-11-28 Thread Michael Welter

I'm getting the following messages when a call is answered by a SIP device:

Nov 28 13:03:01 NOTICE[22824]: rtp.c:1146 ast_rtp_raw_write: RTP 
Transmission error to 192.168.1.254:19262: Operation not permitted


For a Cisco 7940 line, I have the following sip.conf entry:

[desk2]
type=friend
username=desk2
secret=xxx
host=dynamic
dtmfmode=rfc2833
context=international
canreinvite=no
callerid=xxx3034144980
[EMAIL PROTECTED]
nat=yes
qualify=yes
accountcode=xxx
disallow=all
allow=ulaw
allow=g729

The Asterisk system faces the Internet on a public IP.  The phone is 
behind NAT.


Asterisk version is 1.0.7.

What can I do to fix this problem?

Thanks,



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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Tony Hoyle

Bob Goddard wrote:


It's not so bad... you do get access to firmware to all cisco devices
with that, so if you have more than one device it becomes worth it.



And it is also illegal.


Not true - that's the *point* of the more expensive contracts.  They 
cover you for each device that you own (there's probably a limit.. I 
only have 3 or 4 items though).


I recently had to deal with cisco over a router I bought - they had no 
issue handling it under my current contract (in fact I still have one 
open, waiting for an RMA to complete).


Tony
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Tony Hoyle

Bob Goddard wrote:

I guess that should depend as to whether it is hardware or software only.

AFAIK all smartnet are software only... I've never heard of a hardware 
contract.


(actually they're just an account on TAC which has access to certain 
parts of the website - there's no physical part to them at all).


Tony
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Bob Goddard
On Monday 28 Nov 2005 20:41, Tony Hoyle wrote:
 Bob Goddard wrote:
 It's not so bad... you do get access to firmware to all cisco devices
 with that, so if you have more than one device it becomes worth it.
 
  And it is also illegal.

 Not true - that's the *point* of the more expensive contracts.  They
 cover you for each device that you own (there's probably a limit.. I
 only have 3 or 4 items though).

 I recently had to deal with cisco over a router I bought - they had no
 issue handling it under my current contract (in fact I still have one
 open, waiting for an RMA to complete).

You misunderstand. Buying a smartnet contract for a phone does not
give you the right to download software for other hardware. One 
smartnet contract equals only one device covered.
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Bob Goddard
On Monday 28 Nov 2005 20:42, Tony Hoyle wrote:
 Bob Goddard wrote:
  I guess that should depend as to whether it is hardware or software only.

 AFAIK all smartnet are software only... I've never heard of a hardware
 contract.

No, the vast majority of the smartnet contracts are hardware and software
with some of the hardware having a 2 hour turnaround.

 (actually they're just an account on TAC which has access to certain
 parts of the website - there's no physical part to them at all).

Yes, I know.
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Re: [Asterisk-Users] Help connecting Avaya S8700 and Asterisk through H.323 trunk

2005-11-28 Thread Pablo Chacón
Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
Do you have S8700 with Asterisk working? using oh323 channel??
Maybe can help you my S8700 configuration...
My S8700 configuration is:
---
list ip-interfaces clan

IP INTERFACES
  Num
  Skts Net
ON Slot  Code Sfx Node Name/  Subnet Mask Gateway Address Warn Rgn VLAN
  IP-Address
--    --- --- --- ---  --- 
.
y 04A04 TN799  D  CLND04A04   255.255.255.0   10.64.108.254   400  2   n
  10.64.108.132
---

change signaling-group 23   Page   1 of   5
SIGNALING GROUP

 Group Number: 23 Group Type: h.323
   Remote Office? n  Max number of NCA TSC: 0
 SBS? n   Max number of CA TSC: 0
IP Video? nTrunk Group for NCA TSC:
   Trunk Group for Channel Selection: 23
  Supplementary Service Protocol: a  Network Call Transfer? n
 T303 Timer(sec): 10

   Near-end Node Name: CLND04A04 Far-end Node Name: ASTERISK
 Near-end Listen Port: 1720Far-end Listen Port: 1720
Far-end Network Region: 2
 LRQ Required? n Calls Share IP Signaling Connection? n
 RRQ Required? n
 Media Encryption? n Bypass If IP Threshold Exceeded? n

 DTMF over IP: out-of-bandDirect IP-IP Audio Connections? n
IP Audio Hairpinning? n
 Interworking Message: PROGress
 DCP/Analog Bearer Capability: 3.1kHz

-
display trunk-group 23  Page   1 of  19

TRUNK GROUP

Group Number: 23   Group Type: isdn  CDR Reports: y
  Group Name: ASTERISK-H323   COR: 1TN: 1TAC: #23
   Direction: two-wayOutgoing Display? n Carrier Medium: IP
 Dial Access? yBusy Threshold: 255   Night Service:
Queue Length: 0
Service Type: tie   Auth Code? nTestCall ITC: rest
 Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
 Codeset to Send Display: 0 Codeset to Send National IEs: 6
Max Message Size to Send: 260   Charge Advice: none
  Supplementary Service Protocol: a Digit Handling (in/out): enbloc/enbloc

Trunk Hunt: cyclical QSIG Value-Added? n
   Digital Loss Group: 18
Incoming Calling Number - Delete: Insert: Format:
  Bit Rate: 1200 Synchronization: asyncDuplex: full
 Disconnect Supervision - In? y  Out? n
 Answer Supervision Timeout: 0


display trunk-group 23  Page   2 of  19
TRUNK FEATURES
  ACA Assignment? nMeasured: none  Wideband Support? n
 Internal Alert? nMaintenance Tests? y
   Data Restriction? n NCA-TSC Trunk Member:
  Send Name: y  Send Calling Number: y
Used for DCS? n
   Suppress # Outpulsing? nFormat: public
 Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider

 Replace Restricted Numbers? n
Replace Unavailable Numbers? n
  Send Connected Number: n
Network Call Redirection: noneHold/Unhold Notifications? n
 Send UUI IE? yModify Tandem Calling Number? n
   Send UCID? n
 Send Codeset 6/7 LAI IE? y



 SBS? n  Network (Japan) Needs Connect Before Disconnect? n



display ip-network-region 2 Page   1 of  19
   IP NETWORK REGION
  Region: 2
Location: 1   Authoritative Domain:
Name: ** Pool LR VoIP **
Intra-region IP-IP Direct Audio: yes
MEDIA 

[Asterisk-Users] Avaya 4620SW Invalid Subscription-State - Issue

2005-11-28 Thread [EMAIL PROTECTED]
Hello All,
I'm using an Avaya 4620SW with Asterisk, the phone when hooked up to
the network, works for sometime, (I have not actually monitored the
time) maybe 20-30 minutes, after which the phone will still have a dial
tone, but can't dial out or recieve calls. I scanned thru the logs and
found this:
Got SIP response 400 Invalid Subscription-State back from
192.168.202.200 (
192.168.202.200 is the static IP assigned to the Avaya
device).

Thanks



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[Asterisk-Users] Re: Wrong usage of [] in the extension?

2005-11-28 Thread Rusty Dekema
Sorry to reply to myself, but I need to add some information: 

I have been informed (and now understand why) that the [] syntax does
not do what I had in mind here. Is there any syntax that will do it? If
not, I will just create a separate pattern for each NPA, which is
not a big deal, but I am now curious as to whether there is such a
syntax.

Thanks,
RustyOn 11/28/05, Rusty Dekema [EMAIL PROTECTED] wrote:
Hello,

I am trying to set up my dialplan in such a manner that calls to
numbers in the form 1-NPA-NXX- will only go through if the NPA
dialed is a geographical NPA in the continental United States. 

I have collected a list of all NPAs that I want to allow, and have made
the following dialplan entries (below). Calls that match the pattern do
go through, but for each digit that I dial, I get the following error
on the Asterisk console: 

WARNING[7045]: pbx.c:718 ast_extension_close: Wrong usage of [] in the extension

or

WARNING[7045]: pbx.c:699 ast_extension_match: Wrong usage of [] in the extension

If I comment out the three entries below, the errors stop (and of course the calls do not go through). 

I have checked that each dialplan entry is on one line in the file;
there are no accidental carriage returns or newlines present.

Does anyone know what I am doing wrong? I realize that I have a rather
large pattern defined, but it seems to be using correct syntax as far
as I can tell. If there is a better way to do this, or if someone sees
what I am doing incorrectly, could you please let me know? 

Thanks very much,
Rusty



Extensions.conf excerpt: 


exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,1,Playback(local/ding
ding)

exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9
51|952|954|956|970|971|972|973|978|979|980|985|989]NXX,n,Dial(IAX2/[EMAIL PROTECTED]
jet/${EXTEN},60)

exten = _1[201|202|203|205|206|207|208|209|210|212|213|214|215|216|217|218|219|2
24|225|228|229|231|234|239|240|248|251|252|253|254|256|260|262|267|269|270|276|2
81|301|302|303|304|305|307|308|309|310|312|313|314|315|316|317|318|319|320|321|3
23|325|330|334|336|337|339|347|351|352|360|361|386|401|402|404|405|406|407|408|4
09|410|412|413|414|415|417|419|423|425|430|432|434|435|440|443|469|478|479|480|4
84|501|502|503|504|505|507|508|509|510|512|513|515|516|517|518|520|530|540|541|5
51|559|561|562|563|567|570|571|573|574|580|585|586|601|602|603|605|606|607|608|6
09|610|612|614|615|616|617|618|619|620|623|626|630|631|636|641|646|650|651|660|6
61|662|678|682|684|701|702|703|704|706|707|708|712|713|714|715|716|717|718|719|7
20|724|727|731|732|734|740|754|757|760|763|765|769|770|772|773|774|775|781|785|7
86|801|802|803|804|805|806|810|812|813|814|815|816|817|818|828|830|831|832|843|8
45|847|848|850|856|857|858|859|860|862|863|864|865|870|878|901|903|904|906|908|9
09|910|912|913|914|915|916|917|918|919|920|925|928|931|936|937|940|941|947|949|9

Re: [Asterisk-Users] beginner questions

2005-11-28 Thread Fred Blaise
Hello Amir

On Sun, 2005-11-27 at 20:31 -0800, Amir Aziz wrote:
 Dear List Members,
  
 I am trying to setup a small asterisk box. My configure is pretty
 basic for now. my zaptel.conf is as follows 
[ ... ]
 
 6. What other books/links can be helpful in learning this interesting
 software.
I liked Switching to VoIP from oreilly. Very helpful and numerous
references to *.

  
 I thank you all for you help in advance.
  
 Regards,
 Amir Aziz
 
  
 
 __
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Rich Adamson

  I guess that should depend as to whether it is hardware or software only.
  
 AFAIK all smartnet are software only... I've never heard of a hardware 
 contract.

Smartnet comes in serveral different flavors (eg, 24x7, 8x5) and
all of the flavors cover the hardware in addition to the software (at 
least in the US). Been using smartnet since around 1985.

Cisco actually stockpiles replacement hardware in key locations around
the US based on what customers have signed up for, and those key locations
are _not_ necessarily cisco sites. Most are contractual agreements with
selected carriers, and the carrier actually dispactches the replacement
hardware based on TAC orders. That's why they can sell 24x7 smartnet
contracts with 4 hour response times. The local cisco sales office is
not allowed to stockpile anything and, for the most part, will refer you
to the TAC folks for failures, assistance, etc. Resellers frequently
operate a little differently.

The smartnet contracts for the 79x0 phones includes replacement hardware
should the phone fail for any reason.

 (actually they're just an account on TAC which has access to certain 
 parts of the website - there's no physical part to them at all).

If you read the legal stuff that must be acknowledged when downloading
software (under any cisco contract), you can only legally download the
software for the stuff that you have under contract. But, I've never
heard them enforce the web acknowledgement with any company to date.


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[Asterisk-Users] misdn, busy detection

2005-11-28 Thread Denny Schierz
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

hi,

very often, when the caller hangs up the phone, the isdn phone rings
without stopping. It seems, that asterisk does noch check, that the
caller has hang up.

I have this problem between ISDN-ISDN and ISDN-SIP. Is there a
solution for misdn?

cu denny
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (Darwin)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFDi3V4Klzhkqt9P+ARAmnrAJ482h1lr0BSKpZgc5IbOQPiXV2jNgCZAWAR
fm7V58hiK8dZhtj+Y77yNTo=
=RC3h
-END PGP SIGNATURE-
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Re: [Asterisk-Users] SNOM Phones MWI, Hold Retrieve buttons not working with Asterisk v1.2

2005-11-28 Thread Michiel van Baak
On 13:48, Mon 28 Nov 05, Sascha Deri wrote:
 Additionally, the Retrieve voicemail butotn on the phones no longer 
 work.  The MWI (Message Waiting Indicator) lights up, but when you press 
 the button you get Not Found sip:asterisk@ and busy signal.

I have been fighting with the same thing for hours.
The old asterisk sent MWI messages as
[EMAIL PROTECTED]
The new code sends this as [EMAIL PROTECTED]

This can be overwritten by some setting in sip.conf, but I
didn't look into that.
I simply setup an extension:
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})

That did the trick for me.
Hope this helps you.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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[Asterisk-Users] SNOM and 1.0.9

2005-11-28 Thread Joseph Rothstein
Greetings to all,

I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.

Does anyone know for sure if the hint function works properly in 1.0.9?

If anyone has gotten this to work properly under 1.0.9 please post a sample.

I am using:

Exten = xxx,hint,sip/xxx

Extension no. and sip channel are the same.

On the SNOM I have Destination chosen, and the extension on asterisk.

Maybe output the output from the command 'sip show subscriptions' would also
be nice.

Regards,
Joe


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Re: [Asterisk-Users] Newbie requesting help!

2005-11-28 Thread Joao Carlos Mavimbe
Hi.

Things are the same. I would be glad if you could help out.



Regards.
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[Asterisk-Users] Interface Cards that support QSIG

2005-11-28 Thread Cory Andrews
Anyone know of a board, Digium, Sangoma or other, that supports QSIG?  
Only hardware that I have seen that supports QSIG are Vegastream gateways.


Thanks in Advance!

--
Cory J Andrews
Partner / Purchasing
+++
VOIPSupply.com - Everything you need for VOIP
454 Sonwil Drive
Buffalo, NY 14225
+++
tf voice - 800-398-VOIP X22
l voice - 716.630.1555 X22
f - 716.630.1548
e - [EMAIL PROTECTED]
AIM - b2Cory

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Re: [Asterisk-Users] Problem with pulses dialing on asterisk 1.2

2005-11-28 Thread Tom Rymes

On Nov 28, 2005, at 3:00 PM, John Novack wrote:


Cyrille DERORY wrote:

I'm using asteriskathome 2.0 beta 6 (asterisk 1.2) with SIP  
softphones, 7905G cisco SCCP and analog phone( DTMF dialing). All  
is working nice, however when I change DTMF for an analog pulse  
dialing,my analog phone is not working.


I've found the following :

http://www.voip-info.org/wiki-Asterisk+config+zapata.conf
pulse=yes

http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse 
+dialing

pulsedial=yes


Pulsedial=yes works

You should be aware that inserting a w into the dialstring does  
NOT work with pulsedial, however, so if your PSTN connection is a  
little slow, you may get misdials


Remember you need to restart after making the change.
To be really safe, reboot.


I think this might be a problem peculiar to Asterisk @ home, since I  
cannot get my install to accept pulsedial=yes, either. Even though I  
have tried specifying pulsedial=yes in zapata.conf before the  
channel= line and also in zapata-auto.conf, as well as in  
zapata_additional.conf (along with the extension's config), none of  
the above work. (I figured I'd try everywhere, even if it didn't make  
any sense...)


Anyhow, if I connect to the Asterisk console and run zap show  
channel 1, it reports Pulse phone: no even though pulsedial=yes  
is specified. I don't know how [EMAIL PROTECTED]/AMP would have modified zaptel to  
break this, but it is indeed possible that this is an [EMAIL PROTECTED] specific  
problem.


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

Intelligent technology solutions for small businesses.


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Re: [Asterisk-Users] SNOM and 1.0.9

2005-11-28 Thread pdhales
From memory (at a previous installation) you will need a newer version of
Asterisk than 1.09 for the lights to work.

PaulH

- Original Message - 
From: Joseph Rothstein [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, November 29, 2005 8:32 AM
Subject: [Asterisk-Users] SNOM and 1.0.9


 Greetings to all,

 I am trying to get the line lights on a SNOM 320 to work using 'hint' in
 extensions.conf. Unfortunately I have not been able to get it to work
 properly.

 Does anyone know for sure if the hint function works properly in 1.0.9?

 If anyone has gotten this to work properly under 1.0.9 please post a
sample.

 I am using:

 Exten = xxx,hint,sip/xxx

 Extension no. and sip channel are the same.

 On the SNOM I have Destination chosen, and the extension on asterisk.

 Maybe output the output from the command 'sip show subscriptions' would
also
 be nice.

 Regards,
 Joe


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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Tony Hoyle

Bob Goddard wrote:


You misunderstand. Buying a smartnet contract for a phone does not
give you the right to download software for other hardware. One 
smartnet contract equals only one device covered.


News to me... If cisco queried my TAC request of a router on an existing 
contract bought when I only owned a phone I'd believe it, but they 
didn't bat an eyelid.


I see that smartnet is now available from my usual hardware dealer for 
£40 which is pretty damned competetive (since I paid £150 for it 5 
months ago from an 'official' cisco dealer, who then proceeded to mess 
me about for 2 months before getting off their arses and actually 
registering it)... not sure I'd actually get one again though - the 
dealer support has been very good, it's just cisco's end that has sucked.


Tony
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Re: [Asterisk-Users] Help connecting Avaya S8700 and Asterisk through H.323 trunk

2005-11-28 Thread BJ Weschke
On 11/28/05, Pablo Chacón [EMAIL PROTECTED] wrote:
 Hi BJ Weschke, thanks but unfortunately Ip address is the correct one.
 Do you have S8700 with Asterisk working? using oh323 channel??
 Maybe can help you my S8700 configuration...
 My S8700 configuration is:
 ---

 Yes. There's one client I setup where they have this in production. I
don't have access via DSA anymore though to tell you what the sig
group settings are there. Sorry.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Emailed voicemail messages not being deleted

2005-11-28 Thread Dustin Wenz
That appears to have done the trick...I guess I expected some sort of  
warning at the console if I had inadvertently malformed the parameter  
string. It works now though, so it's all good.


Thanks for the help!

- .Dustin Wenz

On Nov 28, 2005, at 2:15 PM, Gonzalo Servat wrote:


On 11/28/05, Dustin Wenz [EMAIL PROTECTED] wrote:

According to the Asterisk wiki, adding the delete=yes option to a
voicemail definition should automatically delete messages after they
are emailed. This is the format that I'm using:
101 = ,First Last,[EMAIL PROTECTED],attach=yes|delete=yes

[snip]

Try:

101 = ,First Last,[EMAIL PROTECTED],,attach=yes|delete=yes

(notice the extra comma after the email address)

I believe the setting that goes in between the empty commas is the
pager email address

Hope this helps.

Cheers,
Gonzalo
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Re: [Asterisk-Users] SNOM and 1.0.9

2005-11-28 Thread Kevin Hanson

Joseph Rothstein wrote:


Greetings to all,

I am trying to get the line lights on a SNOM 320 to work using 'hint' in
extensions.conf. Unfortunately I have not been able to get it to work
properly.

Does anyone know for sure if the hint function works properly in 1.0.9?

If anyone has gotten this to work properly under 1.0.9 please post a sample.

I am using:

Exten = xxx,hint,sip/xxx

Extension no. and sip channel are the same.

On the SNOM I have Destination chosen, and the extension on asterisk.

Maybe output the output from the command 'sip show subscriptions' would also
be nice.

Regards,
Joe


 

I can't tell you if it works in 1.0.9 as I've only used hints in 1.2 
iterations (beta, rc, final), but, my hints have the channel type in 
upper case:


exten = 509,hint,SIP/509.

Don't know if your extract is direct from your dial plan or just the way 
you typed it in the email.  If it is lower case in your dial plan try 
changing it.


In 1.2 there is a CLI command 'show hints', but I don't believe that 
exists pre 1.2.


Cheers,
Kevin
--
Optimacy Communications, LLC
http://www.optimacycomm.com
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Re: [Asterisk-Users] Upgrade Cisco 7910 with Asterisk ?

2005-11-28 Thread Tony Hoyle

Rich Adamson wrote:

If you read the legal stuff that must be acknowledged when downloading
software (under any cisco contract), you can only legally download the
software for the stuff that you have under contract. But, I've never
heard them enforce the web acknowledgement with any company to date.


I've got a TAC case open at the moment on an IOS bug, waiting for more 
info from me (when I get the router back.. been a while now).  They've 
actively encouraged me to download different versions of IOS - and the 
TAC is under my original contract (which I note is now available from my 
favourite hardware dealer for £40... I paid over £150 for it from a 
'real' cisco dealer.).


Tony
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[Asterisk-Users] Comedian Voicemail? PROBLEMS?

2005-11-28 Thread Martin Joseph

Hi,

I am a newbie,  and I am setting up a simple system to share a PSTN 
line with another location.


In the process of setting this up I am also testing the various codecs.

I am only able to get comedian voicemail (ie dialing 1234) to record or 
playback messages if I use the GSM codec?  Is this normal and expected? 
 If I use ulaw or alaw I get either trash noise or an immediate busy 
signal on attempted message playback.


I am running asterisk 1.2 on OSX 10.4.3.

Thanks for any help or ideas, and sorry if this is a tired old question.
Marty

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[Asterisk-Users] Problem with Internet connection

2005-11-28 Thread José Luis Gómez
Hello.
I`m using asterisk 1.0.9 and it`s working fine until I disconect the WAN
interface. Then asterisk doesn`t work fine, doesn`t make any Dial() and
I don`t know where is the problem. When I connect the WAN interface all
start working fine.
I`m also using NAT in the same server.
I don`t know what asterisk is looking for on the internet.

Regards.

-- 

José Luis Gómez
Qualis Information Technology
Av. Rivadavia 2553, PB Of. 43 EP
0342-4565684 int 102
Bs. As. 011-51990896
www.qualis.com.ar
Soporte 0810-8880022
Santa Fe - Argentina

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Re: [Asterisk-Users] SNOM and 1.0.9

2005-11-28 Thread Michiel van Baak
On 08:48, Tue 29 Nov 05, [EMAIL PROTECTED] wrote:
 From memory (at a previous installation) you will need a newer version of
 Asterisk than 1.09 for the lights to work.

on 1.0.9 the lights work.
In this way:
person is on the phone: light is on
Person is not on the phone: light is off

since 1.2 the lights will blink when the phone is running
and above states work the same.
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D

Why is it drug addicts and computer afficionados are both called users?

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Re: [Asterisk-Users] SNOM and 1.0.9

2005-11-28 Thread BJ Weschke
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote:
 Joseph Rothstein wrote:

 Greetings to all,
 
 I am trying to get the line lights on a SNOM 320 to work using 'hint' in
 extensions.conf. Unfortunately I have not been able to get it to work
 properly.
 
 Does anyone know for sure if the hint function works properly in 1.0.9?
 
 If anyone has gotten this to work properly under 1.0.9 please post a sample.
 

 This is definitely a 1.2 only feature. It is not in 1.0.9.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] small office setup

2005-11-28 Thread Joseph Rothstein








Hi Jason,



There are a couple of boxes on the market these days that
have the following ports:



FXO/ISDN line out to PSTN

2 - FXS - analogue phone (or fax)

WAN port for DSL

As well as wfifi.



Fritz WLAN FON box for example.



Quick design:



Asterisk server at HQ. Each remote employee has one of these
boxes. 



Calls in go to Asterisk, and get forwarded via VOIP to a
phone registered via SIP from one of the FRITZ boxes. If there is no SIP
registration, the call could be redirected back out to the PSTN. Calls from an
employee could go out either Asterisk or the PSTN based on dialplan. You would
need either two FXO ports on the Asterisk (or one ISDN port).



All the rest, voicemail, recording, time of day changes,
etc. can all be done in Asterisk.



Good luck,

Joe











Joseph Rothstein, CCIE

Senior Network Engineer









Comcentrixs
GmbH

Landsberger Str. 155 /
House 2

D-80687 Munich

Office: +49 (0) 89 2444
3168 -212

Softphone: +49 (0) 89
2444 3168 - 8451

Mobile: +49 (0) 172 104 3273

Fax: +49 (0) 89 2444 3168
-999

E-mail: [EMAIL PROTECTED]

Web: http://www.comcentrixs.com



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