date:20051031

 This is a old firmware issue, upgrading the phone firmware 
 everything is working ok with the 7960

Sadly, that's the problem at the moment - I can't seem to get hold of new
firmware for love nor money. Even the hunting for firmware on ebay route
yielded zero results when I had a look yesterday.

Regards,

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


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Re: [Asterisk-Users] SCCP support is making good progress

2005-10-31 Thread Sergio Chersovani


Chris Bagnall ha scritto:


Sadly, that's the problem at the moment - I can't seem to get hold of new
firmware for love nor money. Even the hunting for firmware on ebay route
yielded zero results when I had a look yesterday.
 

Buyu the cheapest cisco smartnet contract and you will be able to 
download the phone firmware upgrade


Sergio
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[Asterisk-Users] Call Pickup in [EMAIL PROTECTED]

2005-10-31 Thread Stephen Arulraj

Anyone out there knows how the call-pickup works on [EMAIL PROTECTED] I 
tried *8 and it did not work. Can a IAXs client also me assigned into a 
call-pickup group?


Thanks in advance,
Stephen


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Re: [Asterisk-Users] Fritz!Card PNP behind a Siemens HiPath 3550 as a BRI trunk

2005-10-31 Thread Stephen Arulraj

Anyone out there with experience connecting this? The Siemens HiPath 
3550 comes with 2 BRI (S0) ports built-in and is configured as a BRI 
trunk interface card.


Thanks in advance.

Stephen


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[Asterisk-Users] H323 one way audio using oh323

2005-10-31 Thread mik sib

Hi all,

through oh323 i can register to my gatekeeper and make
and receive calls.

My gatekeeper routes the incoming call as well as the
outgoing.

The problem is simply that i can't ear nothing from my
SIP ipPhones. I can ear my voice from a normal
telephone in my SIP phone but no viceversa.

How can i debug this situation ? I've no errors in the
log or at the asterisk startup.
How to understand what's happening ?
I've tryed different phones also.
any idea ?
thank you very much
Mik


Here's my oh323.conf
 Configuration of OpenH323 channel driver
--
Version: 0.7.3
Listening on address: 10.0.0.253:1720
Gatekeeper used: [EMAIL PROTECTED]
(Registered)
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref. order: ulaw0
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 1 - 2
UDP (RAS) port range: 1 - 2
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: rfc2833
Max number of inbound H.323 calls: 100
Max number of outbound H.323 calls: 100
Max number of simultaneous H.323 calls: 100
Max call rate (ingress direction): 1.00/30
Default language: en
Default music class: default
Default context: voip-h323

doing a call with the ip phone to the outside world
through the gatekeeper

[2]WrapperAPI::h323_make_call: Making call.
[2]WrapH323EndPoint::MakeCall: Making call to
0258115040
[4]WrapH323EndPoint::CreateConnection: Creating a
H323Connection [32066]
[2]WrapH323Connection::WrapH323Connection: Creation of
WrapH323Connection based on user data.
[2]WrapH323Connection::WrapH323Connection: Call is
outgoing.
[4]WrapH323Connection::WrapH323Connection:
WrapH323Connection created.
[3]WrapH323EndPoint::MakeCall: Call token is
ip$localhost/32066
[3]WrapH323EndPoint::MakeCall: Call reference is 32066
[2]WrapH323Connection::OnSendSignalSetup: Sending
SETUP message...
[3]WrapH323Connection::OnSendSignalSetup: Setting
display name 0432281316 Fabio Violino
[3]WrapH323Connection::OnSendSignalSetup: Setting
calling party number test419
[2]WrapH323Connection::OnAlerting: Ringing phone for
0258115040 ...
[3]WrapH323EndPoint::OpenAudioChannel: Direction =
RECODER, Buffer = 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=45)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 45,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel Asterisk for recording using 1x320 byte
buffers.
[3]WrapH323Connection::OnEstablished:
WrapH323Connection [ip$localhost/32066] established
(FastStartDisabled/noH245Tunneling)
[3]WrapH323EndPoint::OnConnectionEstablished:
Connection [ip$localhost/32066] established.
[3]WrapH323EndPoint::GetConnectionInfo:
[ip$localhost/32066] RTP Media:
10.0.0.253:10004-0.0.0.0:0
[3]WrapH323EndPoint::OpenAudioChannel: Direction =
PLAYER, Buffer = 320
[2]WrapH323EndPoint::OpenAudioChannel: Media format:
FrameSize 8, FrameTime 8, TimeUnits 8
[2]WrapH323EndPoint::OpenAudioChannel: Codec info:
FrameRate 160
[2]WrapH323EndPoint::OpenAudioChannel: Packet size:
160
[2]WrapH323EndPoint::OpenAudioChannel: Frames per
packet: 20
[2]WrapH323EndPoint::OpenAudioChannel: LID Codec
G.711-uLaw-64k
[3]WrapH323EndPoint::OpenAudioChannel: The sound
channel with the application is asterisk-oh323(fd=43)
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
initialized.
[4]PAsteriskSoundChannel::PAsteriskSoundChannel:
Object initialized.
[3]PAsteriskSoundChannel::Open: os_handle 43,
mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
160
[3]WrapH323EndPoint::OpenAudioChannel: Opened sound
channel Asterisk for playing using 1x320 byte
buffers.
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Read: Data read [320 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[5]PAsteriskSoundChannel::Write: Written [160 bytes]
[4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
[5]PAsteriskSoundChannel::Write:

[Asterisk-Users] Segfault on latest head 10/31

2005-10-31 Thread gw

Anyone seen this one so far?  Seems to happen in or outgoing, and even
if I just pick up the channel.

09/15 revision works fine, but the 10/31 checkout is doing this
instantly.  All with HEAD zaptel and libpri

Oh and another off topic thing.

Sometimes I have a way of forgetting I have asterisk running, and do a
module unload.  As you can expect, this causes an EIP and kills the
server.  The server will then stay stuck at the EIP, but does anyone
know of a way to do an auto-reboot?  Or shouldn't the zaptel channel
module not be unloadable while asterisk is running?  Sure I know it's my
fault if I do this by accident, but fortunately the server is only 45
mins away.  Would be rough in another state to make that mistake :)

Asterisk Ready.
*CLI -- Starting simple switch on 'Zap/28-1'
Ouch ... error while writing audio data: : Broken pipe
Warning, flexibel rate not heavily tested!
Segmentation fault



Greg
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[Asterisk-Users] Session Border Control

2005-10-31 Thread Luca Baldantoni

Hi!

Is there an available implementation of Session Border Control on Asterisk?

Thanks a lot in advance!

Luca

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[Asterisk-Users] A2Billing

I am getting the following error when I click on create new ratecard

Fatal error: Cannot redeclare display_minute() (previously declared 
in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52) 
in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
fplan.inc on line 27

Is anybody else experincing this?  I s there a way to fix it?

 thanks
 John Fraser
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Re: [Asterisk-Users] Fritz!Card PNP behind a Siemens HiPath 3550as a BRI trunk

2005-10-31 Thread massimo


Hi Stephen, I don't think you can use fritz card
to connect to a Siemens pbx.
You have to use a card that works in NT mode for
exemple a more cheap compatible Bristuff card.
Refer to this page:
http://www.voip-info.org/wiki/view/Asterisk+zaphfc

Bye


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

Sent: Monday, October 31, 2005 11:06 AM
Subject: Re: [Asterisk-Users] Fritz!Card PNP behind a Siemens HiPath 3550as 
a BRI trunk



Anyone out there with experience connecting this? The Siemens HiPath 3550 
comes with 2 BRI (S0) ports built-in and is configured as a BRI trunk 
interface card.


Thanks in advance.

Stephen


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[Asterisk-Users] Tone generator module

2005-10-31 Thread Obelix



Does asterisk have a module for generating tones, or a set of prerecorded GSM
tones, like 1100Hz tones et cetera?

/Obelix


This message was sent using IMP, the Internet Messaging Program.

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Re: [Asterisk-Users] Tone generator module

2005-10-31 Thread Erik

app_milliamp is your friend

Obelix wrote:
 
 Does asterisk have a module for generating tones, or a set of prerecorded GSM
 tones, like 1100Hz tones et cetera?
 
 /Obelix
 
 
 This message was sent using IMP, the Internet Messaging Program.
 
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RE: [Asterisk-Users] Dial with 44 and +44 prefix

2005-10-31 Thread James Steven

Thanks, thought that should work but had a type error which have now
corrected.  One further question, how can I set up a line so that if 440 is
dialled before a number the 0 is taken out so only 44 is actually used?

Thanks again.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: 28 October 2005 12:30
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Dial with 44 and +44 prefix

 exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
 exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})

 How can I configure my
 extensions.conf to dial a number starting with 44 to dial without 
 changes?  Also a number sent from Outlook starting with +44?

exten = _44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

Not sure what format outlook sends its numbers in (i.e. whether it sends +44
or whether it translates it into 0044 before sending it, or possibly even
stripping it off if it knows you're in the UK already)

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited This email is made from 100%
recycled electrons


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RE: [Asterisk-Users] A2Billing

2005-10-31 Thread Sam Tam

Go to their website and download the most up to date version and then try it
again..



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: 31 October 2005 10:43
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] A2Billing

I am getting the following error when I click on create new ratecard

Fatal error: Cannot redeclare display_minute() (previously declared 
in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52) 
in
/var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
fplan.inc on line 27

Is anybody else experincing this?  I s there a way to fix it?

 thanks
 John Fraser
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Re: [Asterisk-Users] Tone generator module

2005-10-31 Thread Obelix

Quoting Erik [EMAIL PROTECTED]:

Where can I download it from? I searched the lists and the web for any reference
to it and there is no mention of it.

Regards

Obelix

 app_milliamp is your friend

 Obelix wrote:
 
  Does asterisk have a module for generating tones, or a set of prerecorded
 GSM
  tones, like 1100Hz tones et cetera?
 
  /Obelix
 
  
  This message was sent using IMP, the Internet Messaging Program.
 
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RE: [Asterisk-Users] Dial with 44 and +44 prefix

 One further question, how can I set up a 
 line so that if 440 is dialled before a number the 0 is taken 
 out so only 44 is actually used?

exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})

You could probably do it by playing around with different offets as well:

exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

This would be more flexible if you wanted to do the same for different
country codes, for example:

exten = _NX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

That would remove the zero from any 2-digit country code.

exten = _NXX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::3}${EXTEN:4})

That'd do the same thing for a 3-digit country code.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Tone generator module

2005-10-31 Thread Eric \ManxPower\ Wieling


Obelix wrote:

Quoting Erik [EMAIL PROTECTED]:

Where can I download it from? I searched the lists and the web for any reference
to it and there is no mention of it.

Regards

Obelix



app_milliamp is your friend

Obelix wrote:


Does asterisk have a module for generating tones, or a set of prerecorded


GSM


tones, like 1100Hz tones et cetera?

/Obelix



show application milliwatt
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Re: [Asterisk-Users] Dial with 44 and +44 prefix

2005-10-31 Thread Marc Storck

To bad that prefixes like +220 (Gambia), +230 (Mauritius), +240 
(Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 (Saint Helena), 
+350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech 
Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 (Timor 
Leste), +680 (Palau), +690 (Tokelau), +800 (IFPS), +850 (Northern 
Korea), +870 (Inmarsat), +880 (Bangladesh) and +960 (Maldives) exist, 
otherwise your example would have worked. But you may always include 
these exceptions into your dialplan.


Regards,

Marc

Chris Bagnall wrote:
One further question, how can I set up a 
line so that if 440 is dialled before a number the 0 is taken 
out so only 44 is actually used?



exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})

You could probably do it by playing around with different offets as well:

exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

This would be more flexible if you wanted to do the same for different
country codes, for example:

exten = _NX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

That would remove the zero from any 2-digit country code.

exten = _NXX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::3}${EXTEN:4})

That'd do the same thing for a 3-digit country code.

Regards,

Chris


--
voipGATE.com
Support Team

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[Asterisk-Users] IAX2 trunks encrypted?

2005-10-31 Thread Stefan Gofferje


Hi folks,

I understand that IAX2 supports public key authentication. Is the 
transmission also encrypted or is it possible to encrypt an IAX2 trunk 
between 2 *s?


Regards,
Stefan

--
 (o_   Stefan Gofferje| SCLT, MCP
 //\   Reg'd Linux User #247167   | VCP #2263
 V_/_  Heckler  Koch - the original point and click interface

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RE: [Asterisk-Users] Dial with 44 and +44 prefix

 To bad that prefixes like +220 (Gambia), +230 (Mauritius), 
 +240 (Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 
 (Saint Helena), 
 +350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech
 Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 
 (Timor Leste), +680 (Palau), +690 (Tokelau), +800 (IFPS), 
 +850 (Northern Korea), +870 (Inmarsat), +880 (Bangladesh) and 
 +960 (Maldives) exist, otherwise your example would have 
 worked. But you may always include these exceptions into your 
 dialplan.

Oops :-)

On a more serious note, in that case it's almost impossible to determine
where the country code ends and the local bit starts, unless, as you say,
every possible extension is defined independently.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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RE: [Asterisk-Users] A2Billing

I believe this is the latest version.

Open_A2Billing_version_Raccoon.tar.gz
as of Oct 30 2005.

On Mon, 31 Oct 2005 11:54:18 -, Sam Tam wrote
 Go to their website and download the most up to date version and 
 then try it again..
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
 Sent: 31 October 2005 10:43
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] A2Billing
 
 I am getting the following error when I click on create new ratecard
 
 Fatal error: Cannot redeclare display_minute() (previously declared 
 in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52) 
 in
 /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
 fplan.inc on line 27
 
 Is anybody else experincing this?  I s there a way to fix it?
 
  thanks
  John Fraser
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[Asterisk-Users] [Asterisk Voicemail] Quota

2005-10-31 Thread MOREIRA carlos

Hello,

Is there a way to put a voicemail quota to a SIP user? I mean a quota on the
user's mailbox instead
of a particular message of the user like the 'maxmessage' does currently.
Quata can be total file size of message or
total minutes of messages of a mailbox.

Thanks for your help 

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Re: [Asterisk-Users] chan_bluetooth and audio problem

2005-10-31 Thread José Luis Gómez

Hello.
I solved my problem. I got by cvs the last chan_bluetooth.c and it
works.
cvs -d :pserver:[EMAIL PROTECTED]:/home/cvs login
Best regards,
  José Luis Gómez

El sáb, 29-10-2005 a las 00:24 +0300, Vlasis Hatzistavrou escribió:
 Hello,
 
 We had similar problems with chan_bluetooth and various mobile devices.
 
 I suppose that chan_bluetooth is in a very early stage. We tried to 
 contact the author of the channel with debugging information etc but 
 without luck...
 
 There is also the chance that the project may be stalled...
 
 Best regards,
 Vlasis Hatzistavrou.
 
 José Luis Gómez wrote:
 
 Hello.
 I'm having problem with motorola v635 and asterisk. I can make a call
 but I can't hear any audio and the other side of the call can hear me
 (one way audio).
 I'm using usb to bluetooth adaptor (noganet).
 I'm using gentoo with kernel 2.6.13-r2, asterisk 1.0.9 and
 chan_bluetooth 0.0.1_pre20050212.
 What's may be wrong?
 
 I show you my files:
 - bluetooth.conf:
 [general]
 interface = 0
 [00:15:A8:A8:19:82]
 name= V635
 type= HS
 channel = 3
 autoconnect = yes
 # If I put channel 7, the other side of the call can't hear me (no
 audio). The audio stay on the phone (I can hear the call on phone).
 
 - hcid.conf
 options {
 autoinit yes;
 security auto;
 pairing multi;
 pin_helper /usr/bin/bluepin;
 }
 device {
 name Asterisk;
 class 0x200404;
 iscan enable;
 pscan enable;
 lm accept;
 lp rswitch,hold,sniff,park;
 }
 
 - rfcomm.conf
 rfcomm0 {
 bind yes;
 device xx:xx:xx:xx:xx:xx;
 channel 7;
 comment motoV635;
 }
 
 Thanks in advance.
   José Luis
 
 
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-- 

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] H323 one way audio using oh323

2005-10-31 Thread Daniel Varella de Oliveira

Mik,

 Your asterisk server is another machine of your GK ?  You can start verifying 
if the traffic between the machines (related to RTP packets) is ok.
 Do you have firewall ?
-- 

[ ]'s

Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
Rio de Janeiro - Brasil
www.tecnologiaip.com.br




On Monday 31 October 2005 08:05, mik sib wrote:
 Hi all,

 through oh323 i can register to my gatekeeper and make
 and receive calls.

 My gatekeeper routes the incoming call as well as the
 outgoing.

 The problem is simply that i can't ear nothing from my
 SIP ipPhones. I can ear my voice from a normal
 telephone in my SIP phone but no viceversa.

 How can i debug this situation ? I've no errors in the
 log or at the asterisk startup.
 How to understand what's happening ?
 I've tryed different phones also.
 any idea ?
 thank you very much
 Mik


 Here's my oh323.conf
  Configuration of OpenH323 channel driver
 --
 Version: 0.7.3
 Listening on address: 10.0.0.253:1720
 Gatekeeper used: [EMAIL PROTECTED]
 (Registered)
 FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
 Supported formats in pref. order: ulaw0
 Jitter buffer limits (min/max): 20-100 ms
 TCP port range: 1 - 2
 UDP (RAS) port range: 1 - 2
 UDP (RTP) port range: 1 - 2
 IP Type-of-Service value: 0
 User input mode: rfc2833
 Max number of inbound H.323 calls: 100
 Max number of outbound H.323 calls: 100
 Max number of simultaneous H.323 calls: 100
 Max call rate (ingress direction): 1.00/30
 Default language: en
 Default music class: default
 Default context: voip-h323

 doing a call with the ip phone to the outside world
 through the gatekeeper

 [2]WrapperAPI::h323_make_call: Making call.
 [2]WrapH323EndPoint::MakeCall: Making call to
 0258115040
 [4]WrapH323EndPoint::CreateConnection: Creating a
 H323Connection [32066]
 [2]WrapH323Connection::WrapH323Connection: Creation of
 WrapH323Connection based on user data.
 [2]WrapH323Connection::WrapH323Connection: Call is
 outgoing.
 [4]WrapH323Connection::WrapH323Connection:
 WrapH323Connection created.
 [3]WrapH323EndPoint::MakeCall: Call token is
 ip$localhost/32066
 [3]WrapH323EndPoint::MakeCall: Call reference is 32066
 [2]WrapH323Connection::OnSendSignalSetup: Sending
 SETUP message...
 [3]WrapH323Connection::OnSendSignalSetup: Setting
 display name 0432281316 Fabio Violino
 [3]WrapH323Connection::OnSendSignalSetup: Setting
 calling party number test419
 [2]WrapH323Connection::OnAlerting: Ringing phone for
 0258115040 ...
 [3]WrapH323EndPoint::OpenAudioChannel: Direction =
 RECODER, Buffer = 320
 [2]WrapH323EndPoint::OpenAudioChannel: Media format:
 FrameSize 8, FrameTime 8, TimeUnits 8
 [2]WrapH323EndPoint::OpenAudioChannel: Codec info:
 FrameRate 160
 [2]WrapH323EndPoint::OpenAudioChannel: Packet size:
 160
 [2]WrapH323EndPoint::OpenAudioChannel: Frames per
 packet: 20
 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec
 G.711-uLaw-64k
 [3]WrapH323EndPoint::OpenAudioChannel: The sound
 channel with the application is asterisk-oh323(fd=45)
 [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
 initialized.
 [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
 initialized.
 [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
 Object initialized.
 [3]PAsteriskSoundChannel::Open: os_handle 45,
 mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
 160
 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound
 channel Asterisk for recording using 1x320 byte
 buffers.
 [3]WrapH323Connection::OnEstablished:
 WrapH323Connection [ip$localhost/32066] established
 (FastStartDisabled/noH245Tunneling)
 [3]WrapH323EndPoint::OnConnectionEstablished:
 Connection [ip$localhost/32066] established.
 [3]WrapH323EndPoint::GetConnectionInfo:
 [ip$localhost/32066] RTP Media:
 10.0.0.253:10004-0.0.0.0:0
 [3]WrapH323EndPoint::OpenAudioChannel: Direction =
 PLAYER, Buffer = 320
 [2]WrapH323EndPoint::OpenAudioChannel: Media format:
 FrameSize 8, FrameTime 8, TimeUnits 8
 [2]WrapH323EndPoint::OpenAudioChannel: Codec info:
 FrameRate 160
 [2]WrapH323EndPoint::OpenAudioChannel: Packet size:
 160
 [2]WrapH323EndPoint::OpenAudioChannel: Frames per
 packet: 20
 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec
 G.711-uLaw-64k
 [3]WrapH323EndPoint::OpenAudioChannel: The sound
 channel with the application is asterisk-oh323(fd=43)
 [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
 initialized.
 [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object
 initialized.
 [4]PAsteriskSoundChannel::PAsteriskSoundChannel:
 Object initialized.
 [3]PAsteriskSoundChannel::Open: os_handle 43,
 mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize
 160
 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound
 channel Asterisk for playing using 1x320 byte
 buffers.
 [5]PAsteriskSoundChannel::Write: Written [160 bytes]
 [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
 [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
 [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]

Re: [Asterisk-Users] lilte help please

2005-10-31 Thread Rich Adamson


 problem i can't get asterisk to dial to sip provider no matter what provider 
 i choose
 
 the prefix and telephone format is the main problem and i cant figure it even 
 thoug i looked at example and 
diD not work for me
 
 i took exmple on nufone and net2phone configs !
 
 IF I UNDERSTAND THINGS WELL, i should dial 9 then phone number !, i always 
 get you dialed worgn number 
 
 any ideas
 [OUTGOING]
 exten = _91NXXNXX,1,Answer()
 
 exten = _91NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
 
 exten = _91NXXNXX,3,Congestion

You do not want to answer a call that is in the calling process.
Remove that.

To provide any better answers, we'll need ot see the context that
your sip phones are in along with any other contexts that are
included.

In your example above for nuphone, do you have a context like [nuphone]?
If so, what statements are included in it?

Can you copy/paste what the CLI is showing when you place a call?
It would be helpful to see that.

Until you understand exactly what you're doing, get rid of the n as
a priority and simply use numeric sequential numbers. In the above
example, change to 91NXXNXX,2,Dial and watch your CLI when placing
a call.


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Re: [Asterisk-Users] Segfault on latest head 10/31

2005-10-31 Thread Rich Adamson


 Anyone seen this one so far?  Seems to happen in or outgoing, and even
 if I just pick up the channel.

Nope. cvs-head from yesterday and the last several days are working
just fine on fc3. What distro are you using?

 09/15 revision works fine, but the 10/31 checkout is doing this
 instantly.  All with HEAD zaptel and libpri
 
 Oh and another off topic thing.
 
 Sometimes I have a way of forgetting I have asterisk running, and do a
 module unload.  As you can expect, this causes an EIP and kills the
 server.  The server will then stay stuck at the EIP, but does anyone
 know of a way to do an auto-reboot?  Or shouldn't the zaptel channel
 module not be unloadable while asterisk is running?  Sure I know it's my
 fault if I do this by accident, but fortunately the server is only 45
 mins away.  Would be rough in another state to make that mistake :)

kernel modules should not be unloaded. The fix for that is remedial
training on behalf of the sys admin (you).
 
 Asterisk Ready.
 *CLI -- Starting simple switch on 'Zap/28-1'
 Ouch ... error while writing audio data: : Broken pipe
 Warning, flexibel rate not heavily tested!
 Segmentation fault

The above generally means you've got something very wrong in your
/etc/zaptel.conf or /etc/asterisk/zapata.conf definitions. Best guess
is the zap channels defined in zaptel.conf don't match those defined
in zapata.conf.


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[Asterisk-Users] FXS Disconnect Supervision (Kewlstart / Open Loop Disconnect)

2005-10-31 Thread David Stude




Hello,

For the past week or 
so, I've been looking everywhere for information about disconnect supervision 
and have come to the following conclusion: there is a plethora of 
information available on RECEIVING disconnect supervision signaling via an FXO 
port on analog cards like Voicetronix and Digium. However, there is very 
little available about providing disconnect supervision signaling via an FXS 
port. 

Here's some general 
(the most comprehensive I've found) information: http://www.sandman.com/cpcbull.html

The upshot of this 
is that I'm trying to connect aNorstar MICS system,which has FXO 
analog ports, to our new Asterisk system, using FXS ports. The Norstar 
only recognizes disconnect supervision and, otherwise, will not free up the line 
unless explicitly told to do so, so it's necessary for me to provide that signal 
from a FXS port on a card that's ready for use in Asterisk. I've already 
used one card and, at least until now, have not had any 
success.

Can anyone provide 
any information or experience regarding this or advise me on another 
solution?

-David 
Stude
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[Asterisk-Users] Timestamps in Console?

2005-10-31 Thread tmassey


Hello!

Lately, I've been keeping a close eye on an Asterisk box by staying logged
into the console for long periods of time. However, it can be very
difficult to know how long a telephone call lasts when this is all you
see:

 -- Executing Dial(SIP/SIP105-8e34,
Zap/g2/Number|60|t) in new stack
  -- Called g2/Number
  -- Zap/5-1 answered SIP/SIP105-8e34
  -- Hungup 'Zap/5-1'

Did that telephone call last only a
few seconds because there was a problem, or a few minutes because there
wasn't? It's impossible to tell.

Is there a way to add timestamps to
each line in the console so you know exactly how long a call took? Or
is there another way of telling directly within the console?

Thank you very much!

Tim Massey
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Re: [Asterisk-Users] FXS Disconnect Supervision (Kewlstart / Open Loop Disconnect)

On Monday 31 October 2005 08:37, David Stude wrote:
 The upshot of this is that I'm trying to connect a Norstar MICS system,
 which has FXO analog ports, to our new Asterisk system, using FXS ports.
 The Norstar only recognizes disconnect supervision and, otherwise, will
 not free up the line unless explicitly told to do so, so it's necessary
 for me to provide that signal from a FXS port on a card that's ready for
 use in Asterisk.  I've already used one card and, at least until now,
 have not had any success.

I interfaced our Norstar MICS trunk lines to a cheap-ass Carrier Access Access 
Bank I with FXS ports and I *never* had any issues with the MICS not dropping 
the trunk line when the call was completed.  I now use a DTI+PRI keycode and 
do the Asterisk--Norstar connection with a PRI but it worked fine in the 
original case.

Basically if you tell Asterisk to use KewlStart signalling on the line it will 
send the CPD bit pattern on to the line, but if the channel bank/remote 
equipment doesn't know what to do with it, nothing will happen.  This is the 
case with the AB1 and AB2.  The Carrier Access Adit600 knows how to sense and 
send CPD so it's not an issue there, and the TDM FXS modules I believe do 
handle the CPD state properly.

Actually I just checked the wctdm driver and it does seem to support polarity 
reversal when instructed from Asterisk, which is what would happen when 
KewlStart signaling is selected.  (wctdm.c, wctdm_ioctl() function, 
ZT_SETPOLARITY case if you're curious.)

Have you actually tried this, or are you raising potential concerns?

-A.
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Re: [Asterisk-Users] Timestamps in Console?

On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote:
 Is there a way to add timestamps to each line in the console so you know
 exactly how long a call took?  Or is there another way of telling directly
 within the console?

Of course it's possible, but you'll be maintaining the patch yourself.  :-)  
Why not just enable logging and watch the logfile?  You'll get full 
timestamps there with each line.

-A.
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[Asterisk-Users] RE: Asterisk to Avaya IP Office

2005-10-31 Thread Clauss, Chris

Title: Re: [Asterisk-Users] TDM01B vs. X100P








On the IP Office, try making sure that
fast start is off on the h.323 trunk links. Also, look in Monitor on the IP
Office, see what errors are coming up.





Kind regards,

Chris Clauss

Avaya Certified Expert; Cisco CCDA; Microsoft MCSE

Strategic Products and Services
AVAYA 2003 Business Partner of the Year

3
Wing Drive
Cedar Knolls, NJ 07927

973-359-8557 Voice
973-944-5800 Fax













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Rahn
Sent: Sunday, October 30, 2005
10:20 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Asterisk to Avaya IP
Office









has anyone had any luck connecting * to IPOFFICE via h323
trunk





I can call * from IPO but don't get a connection the other
way 











the * box is sending packets to the ipoffice I see the
Call hit the IPOFFICE as an H323 event but it doesn't actually
connect a call

















thanks
















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Re: [Asterisk-Users] Timestamps in Console?

2005-10-31 Thread Tzafrir Cohen

On Mon, Oct 31, 2005 at 09:30:58AM -0500, [EMAIL PROTECTED] wrote:
 Hello!
 
 Lately, I've been keeping a close eye on an Asterisk box by staying logged 
 into the console for long periods of time.  However, it can be very 
 difficult to know how long a telephone call lasts when this is all you 
 see:
 
-- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in new 
 stack
 -- Called g2/Number
 -- Zap/5-1 answered SIP/SIP105-8e34
 -- Hungup 'Zap/5-1'
 
 Did that telephone call last only a few seconds because there was a 
 problem, or a few minutes because there wasn't?  It's impossible to tell.
 
 Is there a way to add timestamps to each line in the console so you know 
 exactly how long a call took?  Or is there another way of telling directly 
 within the console?

  tail -f /var/log/asterisk/messages

I figure you should play a bit with the settings in logger.conf to
create a log file that'll contain exactly what you want.

-- 
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] Geneys

2005-10-31 Thread Leandro Tenorio

Are u serius?


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jonathan k. Creasy
 Sent: Friday, October 28, 2005 9:49 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Geneys
 
 Anyone using the Genesys framework with an Asterisk PBX? 
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Re: [Asterisk-Users] Segfault on latest head 10/31

On Monday 31 October 2005 05:11, [EMAIL PROTECTED] wrote:
 Sometimes I have a way of forgetting I have asterisk running, and do a
 module unload.  As you can expect, this causes an EIP and kills the
 server.  The server will then stay stuck at the EIP, but does anyone
 know of a way to do an auto-reboot?  Or shouldn't the zaptel channel
 module not be unloadable while asterisk is running?  Sure I know it's my
 fault if I do this by accident, but fortunately the server is only 45
 mins away.  Would be rough in another state to make that mistake :)

...??  

[EMAIL PROTECTED]:~# rmmod wct4xxp
ERROR: Module wct4xxp is in use
[EMAIL PROTECTED]:~# rmmod zaptel
ERROR: Module zaptel is in use by wct4xxp

I can't remove them when Asterisk is running.

What distro?  lsmod should show a nonzero use count for your zaptel and 
lowlevel hardware driver.

-A.
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Re: [Asterisk-Users] Timestamps in Console?

2005-10-31 Thread tmassey


[EMAIL PROTECTED] wrote on 10/31/2005
08:53:35 AM:

 On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote:
  Is there a way to add timestamps to each line in the console
so you know
  exactly how long a call took? Or is there another way of
telling directly
  within the console?
 
 Of course it's possible, but you'll be maintaining the patch yourself.
:-) 
 Why not just enable logging and watch the logfile? You'll get
full 
 timestamps there with each line.

'Cause I can't do things like show channels?
Yeah, yeah, I could have two windows open, I gues...

Me? Lazy?

Tim Massey
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Re: [Asterisk-Users] no sip peers after restarting asterisk?

2005-10-31 Thread Kevin P. Fleming


Rich Adamson wrote:

Just update cvs-head again at 7:45pm CST. Seems the issue still exists. 
Any thoughts on me opening a bug tracker item on this?


Always a good idea (and cheap too!)
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RE: [Asterisk-Users] FXS Disconnect Supervision (Kewlstart / OpenLoop Disconnect)

2005-10-31 Thread akohlsmith-asterisk

Hi Andrew,

Thanks for responding.  Yes, I noticed that some code *seems* to support
this.

I started with a Voicetronix Openswitch12, which has, even in its driver
code, no support for anything resembling kewlstart.  After figuring out that
the MICS seemed to respond to opening the circuit for a short time, I tried
to program a flash into asterisk (and then tried to write simpler code to
interface with their driver) and found that doing anything with flash or
on/off hook status while the card was in FXS mode made the port totally
unstable.  I've been dealing with someone at Voicetronix and, after the
usual WHY would you use FXS to interface with a PBX?!, he seemed
helpful...until I told him that I figured that there was absolutely nothing
that I could imagine, software-based, that I could do to get around it.

After my boss caught wind of it, he's now trying to control the situation a
bit.  We ordered a Digium card AND now he wants to create a circuit with as
many relays as we have lines and trigger that using signals from the
parallel port.  I'm extremely hesitant to go that route for obvious reasons.
I'm hoping the Digium card works, but if it doesn't, I'm stuck either
returning a lot of hardware or going to get solder burns.

-Dave


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Monday, October 31, 2005 8:51 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] FXS Disconnect Supervision (Kewlstart /
OpenLoop Disconnect)

On Monday 31 October 2005 08:37, David Stude wrote:
 The upshot of this is that I'm trying to connect a Norstar MICS 
 system, which has FXO analog ports, to our new Asterisk system, using FXS
ports.
 The Norstar only recognizes disconnect supervision and, otherwise, 
 will not free up the line unless explicitly told to do so, so it's 
 necessary for me to provide that signal from a FXS port on a card 
 that's ready for use in Asterisk.  I've already used one card and, at 
 least until now, have not had any success.

I interfaced our Norstar MICS trunk lines to a cheap-ass Carrier Access
Access Bank I with FXS ports and I *never* had any issues with the MICS not
dropping the trunk line when the call was completed.  I now use a DTI+PRI
keycode and do the Asterisk--Norstar connection with a PRI but it worked
fine in the original case.

Basically if you tell Asterisk to use KewlStart signalling on the line it
will send the CPD bit pattern on to the line, but if the channel bank/remote
equipment doesn't know what to do with it, nothing will happen.  This is the
case with the AB1 and AB2.  The Carrier Access Adit600 knows how to sense
and send CPD so it's not an issue there, and the TDM FXS modules I believe
do handle the CPD state properly.

Actually I just checked the wctdm driver and it does seem to support
polarity reversal when instructed from Asterisk, which is what would happen
when KewlStart signaling is selected.  (wctdm.c, wctdm_ioctl() function,
ZT_SETPOLARITY case if you're curious.)

Have you actually tried this, or are you raising potential concerns?

-A.
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[Asterisk-Users] Re: VoiceMailMain() in 1.2-beta

2005-10-31 Thread Steven

O'reilly had a book out before the docs team wrote theirs.

-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   --
Leif Madsen [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
On 10/30/05, David Bandel [EMAIL PROTECTED] wrote:
 Have the OReilley book.  Also the new 1.2 book from asteriskdocs.org.

Pt... they're the same book :)

--
Leif Madsen - http://www.leifmadsen.com
http://www.asteriskdocs.org -- Co-Founder
http://www.oreilly.com/catalog/asterisk -- Co-Author
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RE: [Asterisk-Users] Timestamps in Console?

2005-10-31 Thread Sherwood McGowan

You could always just add some

exten =
NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)

type commands in your dialplan to force output of the date
time, and you can even reduce the amount of verbosity to the CLI by using it
liberally to signify events, so you don't have to watch
EVERYTHING.

Sherwod

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: Monday, October 31, 2005 9:31
AMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] Timestamps in Console?
Hello! Lately, I've been keeping a close eye on an
Asterisk box by staying logged into the console for long periods of time.
However, it can be very difficult to know how long a telephone call
lasts when this is all you see: -- Executing Dial("SIP/SIP105-8e34",
"Zap/g2/Number|60|t") in new stack -- Called g2/Number -- Zap/5-1 answered
SIP/SIP105-8e34 --
Hungup 'Zap/5-1' Did that
telephone call last only a few seconds because there was a problem, or a few
minutes because there wasn't? It's impossible to tell.
Is there a way to add timestamps to each
line in the console so you know exactly how long a call took? Or is
there another way of telling directly within the console? Thank you very much! Tim Massey
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Re: [Asterisk-Users] A2Billing

Please download the last release (http://areski.net/a2billing/),
I corrected some bugs and this was one of them.
Rgds, Areski

On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
 I am getting the following error when I click on create new ratecard

 Fatal error: Cannot redeclare display_minute() (previously declared
 in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
 in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
 fplan.inc on line 27

 Is anybody else experincing this?  I s there a way to fix it?

  thanks
  John Fraser
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[Asterisk-Users] How to remove a VM greeting - go back to default Allison message

2005-10-31 Thread Brent Torrenga

So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voicemail, then Allison will
simply say extension blahblah is not available, leave a message. That is
exactly what I want to have happen.

However, at some point after deleting busy.wav and greet.wav, asterisk will
magically (?) generate two new wav files, that are full of silence. What is
going on here, and how do I get the behavior I am looking for?


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918 Voice
219.836.1138 Facsimile 
BEGIN:VCARD
VERSION:2.1
N:Torrenga;Brent;August;Mr.
FN:Brent August Torrenga
ORG:Torrenga Engineering, Inc.
TITLE:Designer
TEL;WORK;VOICE:(219) 836-8918
TEL;WORK;FAX:(219) 836-1138
ADR;WORK:;;907 Ridge Road;Munster;IN;46321-1771
LABEL;WORK;ENCODING=QUOTED-PRINTABLE:907 Ridge Road=0D=0AMunster, IN 46321-1771
URL;WORK:http://www.torrenga.com
EMAIL;PREF;INTERNET:[EMAIL PROTECTED]
REV:20040209T215756Z
END:VCARD
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[Asterisk-Users] Having Meetme call another conference

2005-10-31 Thread Anish Basu

Joining two conferences together over a LAN should be possible, at least
theoretically.  I am not sure how the performance would be over a WAN or the
public internet.  I am currently working on joining two meetme conferences
together using IAX2 trunking and will post my results after the trial.

Anish Basu
Field Systems Engineer
Softel, Inc.
Phone: (732) 705-9202
Cell: (732) 312-6634 

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[Asterisk-Users] Info on beta1 seem to be broke

2005-10-31 Thread James Sizemore

I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends 
DTMF via sip info packets to another beta1 box. The peer is set to 
receive info.  What I get is a click sound and a very very short tone.
Sound like to me that I get the first part of the tone before it is 
captured and put in the info packet, but the gateway never seems to send

the tone, the packet that gets sent looks like this:

--

 -- SIP read from 192.168.117.4:5060:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.117.4:5060;branch=z9hG4bK7bd677b8
From: WIRELESS CALLER sip:[EMAIL PROTECTED];tag=as37dd610c
To: sip:[EMAIL PROTECTED];tag=as3af9dc41
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 120 INFO
User-Agent: ISDN-NET Voip Gateway
Content-Type: application/dtmf-relay
Content-Length: 24

Signal=5
Duration=500

--- 



I know it is not the receiving box messing things up as I get the same
short short DTMF sound on a cisco IAD. Something is wrong with this 
packet but I just can't see it!!! Is there any rtp that gets sent, 
anyone know what the Content-length does?

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Re: [Asterisk-Users] FXS Disconnect Supervision (Kewlstart / OpenLoop Disconnect)

Why'd your mailing software fake your From: to be my email address?

On Monday 31 October 2005 09:35, [EMAIL PROTECTED] wrote:
 I started with a Voicetronix Openswitch12, which has, even in its driver
 code, no support for anything resembling kewlstart.  After figuring out
 that the MICS seemed to respond to opening the circuit for a short time, I
 tried to program a flash into asterisk (and then tried to write simpler
 code to interface with their driver) and found that doing anything with
 flash or on/off hook status while the card was in FXS mode made the port
 totally unstable.  I've been dealing with someone at Voicetronix and, after

Well generally speaking you can't do a damn thing with a channel once a hangup 
is done.  

 After my boss caught wind of it, he's now trying to control the situation a
 bit.  We ordered a Digium card AND now he wants to create a circuit with as
 many relays as we have lines and trigger that using signals from the
 parallel port.  I'm extremely hesitant to go that route for obvious
 reasons. I'm hoping the Digium card works, but if it doesn't, I'm stuck
 either returning a lot of hardware or going to get solder burns.

hahaha yeah that's a direct way to do it but very hackish for sure.

First things first.  With a voltmeter across tip-ring, what do you see when 
Asterisk hangs up?  The port should be in fxo_ks signaling, of course.  You 
may also want to meter out the wctdm_ioctl() function to make sure that the 
POLARITYREVERSAL IOCTL's getting hit.  

I read some more on that original URL you posted, I imagine that the wctdm 
driver drops battery for POLARITYREVERSAL in FCC mode, but you can verify 
this too.  It *shouldn't* be too difficult to change the function of that 
IOCTL, wctdm is actually pretty nicely documented, and the datasheets for the 
SLIC are available.

-A.
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Re: [Asterisk-Users] A2Billing

I added versioning information (version.release) in the application
(ie agi ./a2billing --version).
It will be more easy to know which version, release you downloaded and
check if a new one is available.

# Last release have an ACL user support  also advanced filter to
select the cards.

Rgds, A.

On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
 I believe this is the latest version.

 Open_A2Billing_version_Raccoon.tar.gz
 as of Oct 30 2005.

 On Mon, 31 Oct 2005 11:54:18 -, Sam Tam wrote
  Go to their website and download the most up to date version and
  then try it again..
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
  Sent: 31 October 2005 10:43
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] A2Billing
 
  I am getting the following error when I click on create new ratecard
 
  Fatal error: Cannot redeclare display_minute() (previously declared
  in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
  in
  /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
  fplan.inc on line 27
 
  Is anybody else experincing this?  I s there a way to fix it?
 
   thanks
   John Fraser
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Re: [Asterisk-Users] A2Billing

Hi,

would I have to go through the entire installation again?

 Thanks
John Fraser

On Mon, 31 Oct 2005 09:46:44 -0500, Areski K wrote
 Please download the last release (http://areski.net/a2billing/),
 I corrected some bugs and this was one of them.
 Rgds, Areski
 
 On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
  I am getting the following error when I click on create new ratecard
 
  Fatal error: Cannot redeclare display_minute() (previously declared
  in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
  
in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
  fplan.inc on line 27
 
  Is anybody else experincing this?  I s there a way to fix it?
 
   thanks
   John Fraser
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Re: [Asterisk-Users] How to remove a VM greeting - go back to default Allison message

2005-10-31 Thread Darrick Hartman


Brent Torrenga wrote:

So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voicemail, then Allison will
simply say extension blahblah is not available, leave a message. That is
exactly what I want to have happen.

However, at some point after deleting busy.wav and greet.wav, asterisk will
magically (?) generate two new wav files, that are full of silence. What is
going on here, and how do I get the behavior I am looking for?

  
Along these same lines, is it possible to have each person just record 
their name, then have busy or unavailable setup so that what the caller 
would hear is me_saying_my_name then 
allison_saying_is_unavailable/busy.  Each user should be able to 
record their name instead of a complete busy or unavailable message.


Darrick

--
Darrick Hartman
DJH Solutions, LLC
877.901.3113

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[Asterisk-Users] Rugged VoIP phones for use with asterisk

2005-10-31 Thread asterisk

Is there anyone who knows where to find rugged IP phones?

Rugged in this case means that need to be installed on a ship's deck, so it
must be water resistant, anyway compliant with IP 65 specification
(protected against dust and jets of water).

Regards



++
| Francesco Pellegrini   |
|  Frame Srl |
|  Via Antonio Cantore 62/10 |
|  16149 Genova  |
|  Tel.   +39 010 8680570|
|  Fax.  +39 010 6591413 |
|  Cell.  +348 2237798   |
++



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[Asterisk-Users] Adit 600 and Groundstart

2005-10-31 Thread Doug Lytle


Hey everybody.

I have an Adit 600 that I'm not able to get working properly with 
Groundstart.  The Adit (BootCode 1.23), 1 TDM (Version 6.1.2) and 1 FXO 
card (Version 1.12).


The Adit is setup:  ESF,B8ZS. 


1st port is set as signal gs type voice.
2-8 is setup signal ls type voice.

The FXO card is set as follows:

set 1:1 type voice
set 1:1 signal gs

set 1:2-8 type voice
set 1:2-8 signal lscpd

Connection command:

connect a:1:1 1:1
connect a:1:2-8 1:2-8

I've set up the first port in group 1 and the 2nd port in group 2.

I'm able to dial out on group 2(Loopstart) but not on Group 1 (Groundstart).

Putting a phone on the groundstart line and striking ground, I do get 
dial tone.


I've followed the instructions for grounding of the Adit.

Asterisk setup below:

zaptel.conf:

span=1,1,0,esf,b8zs
fxsgs=1
fxsks=2-8
defaultzone=us
loadzone=us

zapata.conf:

[channels]

switchtype = national
context = incoming
signalling = fxs_gs
busydetect = yes
callprogress = no
group = 2
callgroup=1
cancallforward=no
callreturn=no
echotraining=100
echocancelwhenbridged=yes
musiconhold=epi-cd
jitterbuffers=4

channel = 1

; 
switchtype = national
context = incoming
signalling = fxs_ks
busydetect = yes
callprogress = no
group = 1
callgroup=1
cancallforward=no
callreturn=no
echotraining=100
echocancelwhenbridged=yes
musiconhold=epi-cd
jitterbuffers=4

channel = 2-8


extensions.conf:

ignorepat=9
exten = _91NXXNXX,1,NoOP(${EXTEN})
exten = _91NXXNXX,2,Dial(ZAP/g2/${EXTEN:1})
exten = _91NXXNXX,3,NoOP(${DIALSTATUS})
exten = _91NXXNXX,4,Hangup()

Any suggestions will be welcomed.

Thanks!

Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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[Asterisk-Users] lucent TNT h323/sip config?

2005-10-31 Thread Armand Sulter

Does anyone have an example of a lucent
TNT h323 config to work with asterisk ?
I'd like to use sip but it's not supported in the
TAOS we have, if anyone has TAOS 10.x or later
that would be awsome as well, we have the examples
for a sip config.

thx

- Armand
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[Asterisk-Users] Add Contexts Dynamically

2005-10-31 Thread Aaron Clauson

Hi,

Is it possible to dynamically add contexts to the dial plan in any way?

Extensions can be added from the console and therefore also from MAPI but
their doesn't appear to be anyway to add a new context apart from reloading
the configuration files.

The reason I ask is my dialplan is getting quite large and with about 100
changes a day I'm just getting nervous about continually reloading the whole
thing everytime.

Thanks,

Aaron   


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RE: [Asterisk-Users] Dial with 44 and +44 prefix

2005-10-31 Thread James Steven

Hi 

I have inserted the lines you suggested but Asterisk keeps the 0 when
dialling with all alternatives.  Also, I am unsure what the phrase
${EXTEN::2}${EXTEN:3} does?  Could you explain this abit?

My extensions.conf is:

[default] 
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})
exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

Thanks
James

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall
Sent: 31 October 2005 12:13
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Dial with 44 and +44 prefix

 One further question, how can I set up a line so that if 440 is 
 dialled before a number the 0 is taken out so only 44 is actually 
 used?

exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})

You could probably do it by playing around with different offets as well:

exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

This would be more flexible if you wanted to do the same for different
country codes, for example:

exten = _NX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::2}${EXTEN:3})

That would remove the zero from any 2-digit country code.

exten = _NXX0.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN::3}${EXTEN:4})

That'd do the same thing for a 3-digit country code.

Regards,

Chris
--
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recycled electrons


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Re: [Asterisk-Users] A2Billing

No you won't !!! You just have to copy again the AGI  Web Interface
You don't have to change anything in your configuration files or
in your Database. It should be really fast to do!
FYI - areski.net/a2billing
## (ADMIN UI V1.0.3 - CUST UI V1.0.1 - AGI V1.0.1 - DB V1.0.0)
this will keep you informed of the latest release of the different components.
If there is a Database upgrade, you will find in the ChangeLog file the
SQL to apply to make the evolution.
Or perhaps (if I find the time) I will make an automatic script to make so :P


Rgds,
/Areskaille la canaille

On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
 Hi,

 would I have to go through the entire installation again?

  Thanks
 John Fraser

 On Mon, 31 Oct 2005 09:46:44 -0500, Areski K wrote
  Please download the last release (http://areski.net/a2billing/),
  I corrected some bugs and this was one of them.
  Rgds, Areski
 
  On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
   I am getting the following error when I click on create new ratecard
  
   Fatal error: Cannot redeclare display_minute() (previously declared
   in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
  
 in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
   fplan.inc on line 27
  
   Is anybody else experincing this?  I s there a way to fix it?
  
thanks
John Fraser
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[Asterisk-users] VoiceMail help

2005-10-31 Thread Fabio Montemaggiore

I don't receveid e-mail with voicemail.
When I dial 2 with telephone, Asterisk record message
but don't send a e-mail at the mailbox. Why?
I have configuration this file.



In the voicemail.conf
[general]
attach=yes
format=wav
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
sendvoicemail=yes

[zonemessages]
italia=Europe/Rome|'vm-received' Q 'digit/at' HMP

[101]
100 = 100,100,[EMAIL PROTECTED],,|attach=yes


In the dialplan:
exten = 2,1,Answer
exten = 2,2,Wait(1)
exten = 2,3,VoiceMail(u100)
exten = 2,4,Playback(vm-goodbye)
exten = 2,5,Hangup







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[Asterisk-Users] Re: when is 1.2 being released?

2005-10-31 Thread Doug Meredith

Olle E. Johansson [EMAIL PROTECTED] wrote:

Adam Moffett wrote:
 does anyone know when 1.2 will no longer be beta?
 
The quick answer is: When it's ready for release.

Open Source software doesn't really follow a set agenda. 

I don't think that is an accurate statement.  It is certainly true of
Asterisk, but look for example at Eclipse.  They produce an awesome
plan up front with concrete dates.  Different projects work in
different ways.

Doug
-- 
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SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
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[Asterisk-Users] chan.iax2.c errore

2005-10-31 Thread Fabio Montemaggiore

Why Asterisk show this message?

WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device




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Re: [Asterisk-users] VoiceMail help

2005-10-31 Thread Bruno De Luca

did u set the mailserver?
Bruno.

Fabio Montemaggiore wrote:
 I don't receveid e-mail with voicemail.
 When I dial 2 with telephone, Asterisk record message
 but don't send a e-mail at the mailbox. Why?
 I have configuration this file.



 In the voicemail.conf
 [general]
 attach=yes
 format=wav
 skipms=3000
 maxsilence=10
 silencethreshold=128
 maxlogins=3
 sendvoicemail=yes

 [zonemessages]
 italia=Europe/Rome|'vm-received' Q 'digit/at' HMP

 [101]
 100 = 100,100,[EMAIL PROTECTED],,|attach=yes


 In the dialplan:
 exten = 2,1,Answer
 exten = 2,2,Wait(1)
 exten = 2,3,VoiceMail(u100)
 exten = 2,4,Playback(vm-goodbye)
 exten = 2,5,Hangup







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-- 


 BRUNO DE LUCA
 Tel. +39 02 9350 4780 (102)

 FGA Software
 20017 Rho - Via Puccini, 8

 E-Mail :
[EMAIL PROTECTED]
 Internet:
http://www.fgasoftware.com


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RE: [Asterisk-Users] Dial with 44 and +44 prefix

 I have inserted the lines you suggested but Asterisk keeps 
 the 0 when dialling with all alternatives.  Also, I am unsure 
 what the phrase ${EXTEN::2}${EXTEN:3} does?  Could you 
 explain this abit?

The syntax is {EXTEN:initial offset:length}

So EXTEN:3 chops off the first three digits and has no length, so it goes to
the end of the number.
EXTEN::2 has no offset, so starts from the beginning of the number, but only
has a length of 2 (so you get the first 2 digits)
By putting the 2 together you *should* get the first 2 digits, skip one,
then the rest of the number.

 exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
 exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
 exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})
 exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
 exten = _44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

You're using Voiptalk I see :-)

It may be the sort order of extensions that's catching you out here. If _44.
is matched before _440. then you'll still get the zero in there. Try this
for your last line:

exten = _44N.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
(N should be the same as [1-9] I think)

Regards,

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


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[Asterisk-Users] Re: Dial with 44 and +44 prefix

2005-10-31 Thread Tony Mountifield

In article [EMAIL PROTECTED],
James Steven [EMAIL PROTECTED] wrote:
 Hi 
 
 I have inserted the lines you suggested but Asterisk keeps the 0 when
 dialling with all alternatives.  Also, I am unsure what the phrase
 ${EXTEN::2}${EXTEN:3} does?  Could you explain this abit?
 
 My extensions.conf is:
 
 [default] 
 exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
 exten = _00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
 exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})
 exten = _09XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
 exten = _44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

That last line needs to be:

exten = _44[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

otherwise it could be overriding the _440. line.

Also, Z is a shorter equivalent to [1-9]

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-users] VoiceMail help

2005-10-31 Thread Fabio Montemaggiore

I don't set the mailserver.
What can I do?
I use Debian

Thanks






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[Asterisk-Users] Re: Dial with 44 and +44 prefix

2005-10-31 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Chris Bagnall [EMAIL PROTECTED] wrote:
 
 exten = _44N.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
 (N should be the same as [1-9] I think)

N is [2-9], Z is [1-9]

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] A2Billing

cannot overwrite directory `/var/www/asterisk2/htdocs/./A2Billing_UI
cannot remove `/var/www/asterisk2/htdocs/./A2Billing_UI': Is a directory

Ill just start from scratch.
could you put a version number and date on the web page please.
Might save me and others some trouble

On Mon, 31 Oct 2005 10:37:11 -0500, Areski K wrote
 No you won't !!! You just have to copy again the AGI  Web Interface
 You don't have to change anything in your configuration files or
 in your Database. It should be really fast to do!
 FYI - areski.net/a2billing
 ## (ADMIN UI V1.0.3 - CUST UI V1.0.1 - AGI V1.0.1 - DB V1.0.0)
 this will keep you informed of the latest release of the different 
components.
 If there is a Database upgrade, you will find in the ChangeLog file the
 SQL to apply to make the evolution.
 Or perhaps (if I find the time) I will make an automatic script to 
 make so :P
 
 Rgds,
 /Areskaille la canaille
 
 On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
  Hi,
 
  would I have to go through the entire installation again?
 
   Thanks
  John Fraser
 
  On Mon, 31 Oct 2005 09:46:44 -0500, Areski K wrote
   Please download the last release (http://areski.net/a2billing/),
   I corrected some bugs and this was one of them.
   Rgds, Areski
  
   On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
I am getting the following error when I click on create new ratecard
   
Fatal error: Cannot redeclare display_minute() (previously declared
in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
   
  
in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
fplan.inc on line 27
   
Is anybody else experincing this?  I s there a way to fix it?
   
 thanks
 John Fraser
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[Asterisk-Users] (no subject)

2005-10-31 Thread David LEROY




Hi, 

I seek solution for hotel management and billing solution. but I do not 
know which to choose between Astbill or Asterbill ? if you have council.

Thx 
David
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Re: [Asterisk-Users] CallBack Suggestion

2005-10-31 Thread Musaluke AK


Darren,

An example how to call that callback.agi script? The script iself does 
not have usage info.


Thanks


Anthony


Darren Wiebe wrote:
Hello.  You should not need any special hardware for callback.  You will 
(obviously) need  card to connect your box to the pstn.  Do you have 
something setup with freeradius already?  If not, you could quite easily 
setup something like this with ASTCC.  I have a callback script @ 
www.aleph-com.net/astpp.  Somewhere there.  It is way more complicated 
than you need but you can cut out all the user interaction stuff.


Darren Wiebe
[EMAIL PROTECTED]

Abdul Lateef wrote:


Hi friends,

I am new in asterisk, i came for CallBack purpose, i
read from Voip-info.org aboue callback with asterisk
and i am near to collect all information about to
start developing callback system.

Just i have a samall question, Is Callback needs some
special hardware? i have my PSTN phone number i want
to call this number after two ring the call will be
disconnect and the Callback will start to call back to
the caller ID and it should prompt to enter pin id
which will authunticate via freeradius.if the
authuntication is valid it will give some beep for
dialing the international number.

Any kind of suggestion will be hearty appriciated.





Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com


   
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Re: [Asterisk-Users] Timestamps in Console?

2005-10-31 Thread Mark Hulber


Yes, or this for example:

[macro-rhangup]
 
  exten = s,1,NoOp(DIALSTATUS=${DIALSTATUS})

  exten = s,n,NoOp(TIME=${DATETIME})
  exten = s,n,Hangup

I also output the date and time prior to dialing out.

MARK.

Sherwood McGowan wrote:

You could always just add some
 
exten = NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)
 
type commands in your dialplan to force output of the date time, and 
you can even reduce the amount of verbosity to the CLI by using it 
liberally to signify events, so you don't have to watch EVERYTHING.
 
Sherwod



*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of
[EMAIL PROTECTED]
*Sent:* Monday, October 31, 2005 9:31 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Timestamps in Console?


Hello!

Lately, I've been keeping a close eye on an Asterisk box by
staying logged into the console for long periods of time.
 However, it can be very difficult to know how long a telephone
call lasts when this is all you see:

   -- Executing Dial(SIP/SIP105-8e34, Zap/g2/Number|60|t) in
new stack
-- Called g2/Number
-- Zap/5-1 answered SIP/SIP105-8e34
-- Hungup 'Zap/5-1'

Did that telephone call last only a few seconds because there was
a problem, or a few minutes because there wasn't?  It's impossible
to tell.

Is there a way to add timestamps to each line in the console so
you know exactly how long a call took?  Or is there another way of
telling directly within the console?

Thank you very much!

Tim Massey



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Re: [Asterisk-Users] Call Transfer problems-am I missing something?

2005-10-31 Thread alex

Hi,

Thanks for the clarification.  I had seen that the two options 
existed, but the docs for the dial() command didn't state the 
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
 On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  Hi All,
 
  Recently got call-transfer somewhat working on my asterisk-1.0.9
  install, and came across an interesting problem.  I have an account on a
  VOIP Provider (voipbuster using iax to be exact) and use a line like
  this in extensions.conf to have it handle all outgoing calls beginning
  with 1:
  exten = _1NN,1,Dial(voipbuster/00${EXTEN},t)
  When I call someone and press # on the phone ( I've tried this with
  various softphones and a regular phone connected to a linksys pap2)
  Nothing happens.However, if the called party presses # they get the
  extension prompt, and can then transfer me to an other extension.  Does
  anyone know why the calling party can't initiate the transfer? am I
  missing something?
 
 Yes.  The ,t  in the Dial() options is for callee, the T is for
 caller.  ,tT is for both.
 
 Ciao,
 
 David A. Bandel
 --
 Focus on the dream, not the competition.
 - Nemesis Air Racing Team motto
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Re: [Asterisk-Users] .conf file syntax checker (WAS: VoiceMailMain() in 1.2-beta)

2005-10-31 Thread Anthony Rodgers

I have a codeless language module for BBEdit, if anyone's interested  
- it's not complete yet (I'm adding to it as I go along), but I will  
post it to the wiki, if I could just figure out where it should  
go...


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp


On 30-Oct-05, at 7:18 AM, Rich Adamson wrote:



 using the CLI in - mode showed the problem.  Apparently, I can't
 spell (or I can, but when I was typing, I transposed two letters and
 made it vm-recieved vice vm-received).

 Perhaps a good enhancement would be a syntax checker for the various
 .conf files.

Been there... sure wish telnet/putty had a spell checker. ;)


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RE: [Asterisk-Users] Rugged VoIP phones for use with asterisk

2005-10-31 Thread Ted Gibson

you should use an analog made for ships with a ATA.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 31, 2005 7:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Rugged VoIP phones for use with asterisk


Is there anyone who knows where to find rugged IP phones?

Rugged in this case means that need to be installed on a ship's deck, so it
must be water resistant, anyway compliant with IP 65 specification
(protected against dust and jets of water).

Regards



++
| Francesco Pellegrini   |
|  Frame Srl |
|  Via Antonio Cantore 62/10 |
|  16149 Genova  |
|  Tel.   +39 010 8680570|
|  Fax.  +39 010 6591413 |
|  Cell.  +348 2237798   |
++



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[Asterisk-Users] Calling Name Not Displayed On Incoming

2005-10-31 Thread OTR Comm

Hello,

I am using Cisco 7940/7960 phones and can not get the calling name to
display on incoming calls.  The names and numbers do display in the
Missed Calls and Received Calls menus, but not on incoming.  The caller
id number displays fine on incoming, just not the name.  Anybody know
what might be causing this?

Thanks,
Murrah
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[Asterisk-Users] Release of Asterisk 1.2

2005-10-31 Thread gorand

Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.


Thanks.




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[Asterisk-Users] lilte help please

2005-10-31 Thread Kevin Scott

In his Outgoing context, should it not be 9|1NXXNXX, to strip the 9 from
being sent to the provider?

Kevin

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: October 31, 2005 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] lilte help please

 problem i can't get asterisk to dial to sip provider no matter what
provider i choose

 the prefix and telephone format is the main problem and i cant figure it
even thoug i looked at example and 
diD not work for me

 i took exmple on nufone and net2phone configs !

 IF I UNDERSTAND THINGS WELL, i should dial 9 then phone number !, i always
get you dialed worgn number 

 any ideas
 [OUTGOING]
 exten = _91NXXNXX,1,Answer()

 exten = _91NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})

 exten = _91NXXNXX,3,Congestion

You do not want to answer a call that is in the calling process.
Remove that.

To provide any better answers, we'll need ot see the context that
your sip phones are in along with any other contexts that are
included.

In your example above for nuphone, do you have a context like [nuphone]?
If so, what statements are included in it?

Can you copy/paste what the CLI is showing when you place a call?
It would be helpful to see that.

Until you understand exactly what you're doing, get rid of the n as
a priority and simply use numeric sequential numbers. In the above
example, change to 91NXXNXX,2,Dial and watch your CLI when placing
a call.

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[Asterisk-Users] Caller ID to SIPURA-1000, 2000, 3000 Handset Prlblem in only showing the destination callerId

2005-10-31 Thread Ben Higley

Is there a resolution to this problem. It was posted a few weeks back.
But just chiming in again to see if someone has had any luck:

Problem:
Incoming call to a Sipura 2000, 1000, 3000 ATA.

I use the SetCallerID(name)=blah blah blah
SetCAllerId(number)=1234567890

However, On the handset, in this case, it's extension 2000. I see on the
handset display

Blah Blah Blah
2000

I have tried in the dial command to use the o - as shown in the show
application dial - to use old style, but this has not solved the issue.

This makes the end user not able to go through the caller id missed calls
to dial that person back...

Thanks...


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[Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana

2005-10-31 Thread Walter Willis

tengo un asterisk, alguien conoce algun proveedor que brinde el
sistema de linkar mi asterisk a su servicio para tener tarifa plana a
eeuu.

para llamar por 4 conexiones al miamo tiempo desde mi asterisk?

me parece haber visto que se configuraba con una troncal iax2



2005/10/31, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Does anyone know where the official release of Asterisk 1.2 is? Do we havea time-frame of when this version will be released and how much longer itwill be in BETA.Thanks.___
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Re: [Asterisk-Users] A2Billing