date:20031106

Hi,


> - Original Message - 
>From: marin blu
>To: [EMAIL PROTECTED]
>Sent: Thursday, November 06, 2003 8:19 AM
>Subject: [Asterisk-Users] IAX/SIP Client


>Hi,

 >Do you know if there is a IAX/SIP client to work from an internet browser
?

>MarinBlu

DIAX will ve available as an Active X too which can be integrated in a web
page, but in a future release.

BR,
Dan

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[Asterisk-Users] Festival patch

2003-11-06 Thread Jason Penton

Hi all

Just a note:

The patch for festival is reported to be for version 1.4.2. I have
tested the patch on the latest version of festival (1.4.3) and
everything works fine

Cheers
Jason

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dan
Sent: 06 November 2003 12:00 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Error in app_voicemail2.so after CVS
update

Hi,

- Original Message - 
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 05, 2003 10:05 PM
Subject: Re: [Asterisk-Users] Error in app_voicemail2.so after CVS
update


> On Wednesday 05 November 2003 12:03, Dan wrote:
> >  [app_voicemail2.so]WARNING[1074412256]: File loader.c, Line 232
> > (ast_load_resource): /usr/lib/asterisk/modules/app_voicemail2.so:
> > undefined symbol: ast_localtime
> > WARNING[1074412256]: File loader.c, Line 400 (load_modules):
> > Loading module app_voicemail2.so failed!
>
> I don't know what you're doing wrong, because I just checked out CVS
> and tried this, and it works fine.  ast_localtime() is located in the
> stdtime subdirectory of asterisk.

Those are the involved files :
-rw-r--r--1 root root80346 Nov  5 19:30 app_voicemail2.c
in ./apps
-rw-r--r--1 root root37529 Oct 30 20:13 localtime.c
in ./stdtime

in localtime.c this is the only place where ast_localtime is present:

struct tm *
ast_localtime(timep, p_tm, zone)
const time_t * consttimep;
struct tm *p_tm;
const char * const  zone;

Any other idea?

Thanks,
Dan


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Re: [Asterisk-Users] New IAX software phone (for WIndows platform)

HI,

> When I first load the Gui, I get to see directory displayed on the right
> hand (bottom) below RX/X/TX etc info.
You mean first 12 memories..

> Once a number is dialied, this place is used for Volume control but it
would
> be nice to see the same directory all the time so that dialing would be
> easier.
In order to keep the window as small as possible and still have the full
functionality, the form is changed depend on current status.
When you're in a call, you can not dial another number, so this is the
reason that the memories are not available.

> Did you think about keeping the Dial and Hangup buttons as permanent
feature
> instead of switching back and forth? Sometimes, I also see a delete button
> which may not be necessary?
You must take a closer look at the functionality. The two function buttons
depend (again) on the cuirrent status.
They can display DIAL/DELETE, HANGUP/-, REDIAL/-, REJECT/-
When you start enter a number using keypad, after the first digit you et the
DELETE button too, in order to be able to correct the numbet. Each click on
this button deletes the last digit.
When the number is dialed from the memory, DELETE button delete the whole
number.
I cannot imagine something simpler than that...

>
> Anyway, that is my feedback so far. i will try using some more functions
and
> let you know if I find something strange.
Please check it closer. I still work on the detailed help file which will be
available during the weekend.

> By the way, I had liked the idea of multi-line phone as a drop-down action
> as suggested by someone earlier. Default could be a single line with basic
> features.
In this moment, the phone is very close in functionality with a full
featured single line analog phone (with callerid, callwaiting, callwating
callerid, and so on)
I do not see an imediate reason to have more than one line for a standar
phone.
Do you?

>
> Keep it up.
>
I'll do it...

Thanks for your feedback and best regards,
Dan

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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Jonathan Hogg

On 6 Nov 2003, at 04:32, Tilghman Lesher wrote:

OK, let me get this straight.  Because the Asterisk voicemail menu is
fault tolerant and lets you undo a delete, it's therefore unacceptable.
I don't think the OP said it was unacceptable, just that it wasn't as 
configurable as they would like and they considered that a con. I can 
sympathise - the voicemail system is complicated.

I can punch through all the messages in my mobile phone voicemail with 
three keys. 1 plays the message again, 2 saves it, 3 deletes it. If I 
save or delete a message it automatically advances to the next one. In 
comparison, the Asterisk voicemail program is a dog. Having complex 
functionality is fine as long as the basic functionality isn't made 
obscure.

And before you accuse me of being unable to handle moderately complex 
systems as well. The point the OP was making is that it's not 
*configurable* not that it's too hard. If I choose to have a simpler 
system - or more importantly choose for all the users at an 
installation to have a simpler system - I can't do that.

Jonathan

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[Asterisk-Users] Voicemail2 vs Voicemail

2003-11-06 Thread Rich Adamson


>> Wouldn't that break everybody's dialplans where they would have to
>> replace all occurrences of Voicemail2 with Voicemail and all
>> occurrences of Voicemailmain2 with Voicemailmain?
>
> No, we would register with both names.

Is it necessary (with reasonably current cvs) to make any changes in the
*.conf files to use Voicemail2, or is that happening automatically?



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Re: [Asterisk-Users] archives gsm of asterisk ???

2003-11-06 Thread Olle E. Johansson

Javier Rios wrote:
I want to pass the archives of the voicemail of the a Spanish
They can say to me that software I can use to create the archives gsm 
..
Hello Javier,
If I understand you correctly, you want to translate the sound files
into Spanish?
There is information on the WIki about this, start with the page
on "Asterisk sound files" or the FAQ.
http://www.voip-info.org/tiki-index.php?page=Asterisk
Regards,
/Olle
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RE: [Asterisk-Users] iconnect

I will also need to get an incoming number, which is more money, before I'm
satisfied with outgoing calls

But thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson
Sent: Thursday, November 06, 2003 4:08 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] iconnect

I suspect your VPN is giving cruddy performance.  Try this
experiment:

Call the HQ using the Iconnect service.  This will
cut your VPN out of the loop.  I notice in your
sip.conf you don't register with Iconnect.  You will
have do that so you can accept incomming Iconnect
calls. 

--- Shoval Tomer <[EMAIL PROTECTED]> wrote:
> Hi,
> 
> I was able to connect asterisk to iconnect's service.
> 
> It took me almost two hours, but it's because I was having NAT
> trouble.
> 
> I finally discovered that you can set the iconnect host to
> natrealy.deltathree.com to make it work.
> 
>  
> 

=
Chris Albertson
  Home:   310-376-1029  [EMAIL PROTECTED]
  Cell:   310-990-7550
  Office: 310-336-5189  [EMAIL PROTECTED]
  KG6OMK

__
Do you Yahoo!?
Protect your identity with Yahoo! Mail AddressGuard
http://antispam.yahoo.com/whatsnewfree
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[Asterisk-Users] Re: Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Cees de Groot

Chris Albertson  <[EMAIL PROTECTED]> said:
>Yes it would be nice if someone could port Asterisk to Sun SPARC
>hardware then it could run on Sun's telco-grade Netra boxes
>
AFAIK IBM has telco-grade xSeries boxes. 

-- 
Cees de Groot   http://www.tric.nl <[EMAIL PROTECTED]>
tric, the new way   helpdesk/ticketing software, VoIP/CTI, 
web applications, custom development

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[Asterisk-Users] Re: Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Cees de Groot

Steve Underwood  <[EMAIL PROTECTED]> said:
>I have typically found Linux and even SCO Openserver on x86 servers have 
>better up time than the fully redundant machines from Stratus. Their 
>hardware may not fall over, but their OS does. When it does it takes 1 
>to 2 hours to reboot.
>
Yup. Probably the biggest two sensitive items are disks and fans. The
latter can easily be redundanized and are in any well-designed
rack-mount box, and by employing SCSI disks or at least the more
expensive IDE disks, you can get good scores there as well. 

On our 14 server machines we've had 4 disk failures (only 1 of them
in a non-RAID, so that was downtime - but we keep spares, so that was
only a couple of hours) and zero fan failures since jan 2001, when
they were taken into production. Of course, the hosting environment
(21.5 deg. centigrade, controlled humidity, filtered air) will help a
lot here. I can imagine getting 96% if you put your server in an office
environment where people smoke and the cleaning lady now and then bumps
against it with the vacuum cleaner. YMMV.

Most downtime has been due to OS-level issues. And that's because we're
trying to squeeze too much out of too little hardware ;-). The boxes
that have been doing just one or two things typically go down only for
an OS upgrade. 

-- 
Cees de Groot   http://www.tric.nl <[EMAIL PROTECTED]>
tric, the new way   helpdesk/ticketing software, VoIP/CTI, 
web applications, custom development

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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Olle E. Johansson

Steve Underwood wrote:

And is not based on any standards!

100% of all voicemail systems are not based on standards. There *are* no 
standards for voicemail. There aren't even many common practices. The 
From the SER admin's manual:
---
5.3. Voicemail
5.3.1. Introduction
The voicemail system provides ser with voice announcement and recording capabilities.
Voice messages may then be mailed to the called user. The system relies on ser for '
implementing the SIP stack and communicate with it through FIFO.
It implements the dialog and media handling as described in RFC 3264 (An Offer/Answer 
Model
with the Session Description Protocol) and RFC 1889 (Real time transport protocol)
to realize its goal.

I don't know what RFC 3264 is, but surely Asterisk Voicemail supports RTP ;-)
Otherwise, all I can think of is suggestions to use IMAP for voicemail message stores.
There's been some work within the IETF for voice profiles for Internet Mail.
http://www.ietf.org/html.charters/vpim-charter.html

From the charter:
---
The Voice Profile for Internet Mail (VPIM) Version 2 is currently a
Proposed Standard (RFC 2421) Applicability Statement. It is an
application of Internet Mail originally intended for sending voice
messages between voice messaging systems.  As such, VPIM imposes
several restrictions on the message and transport to support the
characteristics of voice messaging. Many voice mail vendors have
implemented systems according to RFC 2421 and are in the process of
deploying these systems around the world. Most vendors have completed
(or are currently involved in) interoperability testing of VPIM
products and have posted their results on the VPIM website.  This
working group will promote the advancement of VPIM v2 on the standards
track.

/O

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RE: [Asterisk-Users] Ping AGI Demo

Great. And if you don't mind us guys who'll use it to prove asterisk's
working to management, it's even better.

Any chance on adding a dns capable script?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Thursday, November 06, 2003 7:49 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ping AGI Demo

I have a ALPHA version of my new ping AGI demo available.

Access via:
  IAXTel 1-700-923-3645
or
  Dial(IAX2/[EMAIL PROTECTED])

When asked for an extension, enter 2101. This will bring you to the
System Services menu.  The Cepstral version of the ping is option 28,
the Festival version of the ping is option 32.

Please report problems and/or issues directly to me.  I'm trying to get
ping.agi stable enough to release the source code for it under GPL.

Thanks in advance,

Eric aka ManxPower

-- 
Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/

BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

Steve Underwood wrote:

96% uptime would mean nearly 4 hours per month down. I have never 
experiemced anything that bad using the nastiest crappiest no-name 
server parts. unless you want to make a point, like some authors 
do. Then you say the hard disk failed and it took a week to get and 
install a new ones, so the downtime was 24x7 hours. In reality, if 
your service support doesn't stock all the important bits for quick 
replacement, it provides no service at all.

I have typically found Linux and even SCO Openserver on x86 servers 
have better up time than the fully redundant machines from Stratus. 
Their hardware may not fall over, but their OS does. When it does it 
takes 1 to 2 hours to reboot.

Regards,
Steve

Steve,

Like I said this was from memory and so may have been inacurate.. I 
probably should have looked it up before hand..

As it turns out it should probably have been 99.6% availablility.. A 
quick look on Google came up with the following details..

http://www1.us.dell.com/content/topics/global.aspx/power/en/ps1q02_graham?c=us&cs=555&l=en&s=biz

Sorry for any stress caused..

Later..

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RE: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Paul Crick

> And before you accuse me of being unable to handle
> moderately complex systems as well. The point the
> OP was making is that it's not *configurable* not
> that it's too hard. If I choose to have a simpler
> system - or more importantly choose for all the
> users at an installation to have a simpler system
> - I can't do that.
Yeah, there was talk a while back about the whole user interface thing for
Comedian Mail and especially how it works compared to other systems, what
could be done to make migration easier for the users etc. There were some
good threads, check the archives, but chances are not much has changed.

I guess you could write your own voicemail system, use AGI or dive headlong
in to changing the existing Voicemail2 app if you're in to the C thing.

Ultimately it would be really nice if there was an Asterisk flavoured
voicemail that could also be totally configured the way you want it,
including a bunch of sample configs for other legacy voicemail systems,
Octel, Meridian Mail etc..

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Re: [Asterisk-Users] Apple implementation

2003-11-06 Thread John Todd

I am new to Asterisk and Digium card implementation issues. My VAR 
is strongly recommending using Apple hardware and Yellow Dog Linux 
for my telephony project, because of his familiarity with this OS. 
Is the PowerPC an appropriate and stable hardware platform for 
Digium/Asterisk development?

Charles Hatchette
[EMAIL PROTECTED]
I seem to recall in the last eight months that someone declared Macs 
would work with Digium hardware.  Search the -users and -dev archives 
for "yellow dog" or PPC to see what you get.

JT
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[Asterisk-Users] which codec will be used ?

2003-11-06 Thread Peter Hudec

hello,

my situations is as follows.
In our comapny we are planing to have *. I'm testing it now.
If we will buy G729 codec for * ...

UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <-> 
(H323)AS5300 <-> world

the following equipment speeks G729: *, GK ,AS5300.
All call from UA to another endpoing go through *, because of mixed SIP 
and H323 sihnalization.

If the UA does not known G729 (known GSM, G711), which codec wil be used 
between * and AS5300 ? Will * translate GSM codec (or other) to the G729.
If yes, how to get it work?

best regards
hudecof
--
mail: [EMAIL PROTECTED] www: [http://www.postel.sk]
cellular: [+421 02 50203166] icq: [99518783]
gpg: [http://hudecof.net/data/hudecof.gpg]
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Re: [Asterisk-Users] Error in app_voicemail2.so after CVS update

Hi,

- Original Message - 
From: "Dave Cotton" <[EMAIL PROTECTED]>
To: "Asterisk List" <[EMAIL PROTECTED]>
Sent: Wednesday, November 05, 2003 11:55 PM
Subject: Re: [Asterisk-Users] Error in app_voicemail2.so after CVS update


> ...
> In the subject a CVS update was mentioned, you said you did a checkout.
> This may be the same problem as I've just answered in another post with
> term_color etc.  CVS update, for whatever reason, may not have updated
> correctly, I don't know that that is the case, but two posts citing the
> same type of problem...

Thanks a lot.
I have deleted all the source tree and then 'cvs checkout' again and 'make
clean'.
Now everything works as expected.

Best regards,
Dan


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Re: [Asterisk-Users] Apple implementation

2003-11-06 Thread Iain Stevenson

I have been running asterisk on an old PowerMac 9600 and YellowDog Linux 
for about a year now.  Asterisk software builds fine most of the time - 
there seem to be some trivial issues with the Makefiles for codecs at the 
moment.

I have an X100P card as the PSTN interface.  I suspect that the interface 
cards are likely to be your biggest problem - drivers supporting big endian 
systems are needed.  I don't know whether all the Digium drivers do.  ISDN 
cards from AVM and Eicon are not suitable for PPC Linux.

 Iain

--On Wednesday, November 5, 2003 9:17 am -0700 Charles Hatchette 
<[EMAIL PROTECTED]> wrote:

I am new to Asterisk and Digium card implementation issues. My VAR is
strongly recommending using Apple hardware and Yellow Dog Linux for my
telephony project, because of his familiarity with this OS. Is the
PowerPC an appropriate and stable hardware platform for Digium/Asterisk
development?
Charles Hatchette
[EMAIL PROTECTED]




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RE: [Asterisk-Users] archives gsm of asterisk ???

Olle.

I've been in the mailing list for a couple of weeks now.
Many threads are answered with links to your wiki.

Cause of the DNS problem I can't get there, no matter what.

Till this is resolved, are you able to provide me (and many others) the
legit IP address for the web server for me to put in my hosts file?

thanks

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, November 06, 2003 10:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???

Javier Rios wrote:
> I want to pass the archives of the voicemail of the a Spanish
> They can say to me that software I can use to create the archives gsm 
> ..
Hello Javier,
If I understand you correctly, you want to translate the sound files
into Spanish?
There is information on the WIki about this, start with the page
on "Asterisk sound files" or the FAQ.
http://www.voip-info.org/tiki-index.php?page=Asterisk

Regards,
/Olle


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[Asterisk-Users] 6.0 image for Cisco 7960's?

2003-11-06 Thread John Todd

Has anyone managed to get their hands on a 6.0 image for their 7960's 
yet?  Or is it still in beta?

Rumor (official rumor, from Cisco) is that it supports paging and 
intercom.  I'm anxious to start working with those features, if 
they've been implemented sanely.  What would be just as nice would be 
NOTIFY messages for pushing XML URL's to the phones, but sadly that 
feature request has gone uncommented-upon by Cisco, so I will assume 
the worst...

JT
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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

On Wed, 2003-11-05 at 20:41, Peter Brown wrote:
> Gavin,
> 
> So you want a few reasons why you shouldn't use asterisk,
> I can think of some:
> 
> Don't want to use a reliable operating system (linux)

[...]

> Is this sufficient Gavin?

Nice one - I'm already a strong Open-Source pedant, and our business is
built on Apache/Perl/MySQL, etc. 

I'm quite aware of these points (most of them are already included in my
'Pros' section)... this was really only to try and pre-empt the kinds of
questions the mgmt will ask, and I can't produce a recommendation
without enev mentioning the 'bad points' :)

Cheers,
Gavin.

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[Asterisk-Users] which channel format number is right?

2003-11-06 Thread Thomas Haeger

Hi all,

if i enter a "show codecs" at cli * response with:

  1 (1 <<  0)  G.723.1
  2 (1 <<  1)  GSM
  4 (1 <<  2)  G.711 u-law
  8 (1 <<  3)  G.711 A-law
 16 (1 <<  4)  MPEG-2 layer 3
 32 (1 <<  5)  ADPCM
 64 (1 <<  6)  16 bit Signed Linear PCM
128 (1 <<  7)  LPC10
256 (1 <<  8)  G.729A audio
512 (1 <<  9)  SpeeX
   1024 (1 << 10)  iLBC
  65536 (1 << 16)  JPEG image
 131072 (1 << 17)  PNG image
 262144 (1 << 18)  H.261 Video
 524288 (1 << 19)  H.263 Video

If i enter a "show channel" * response with

   Name: H323/ip$XXX.XX.XX.XX:3520/25650
   Type: H323
   UniqueID: 1068111809.487
  Caller ID: <0109901>
DNID Digits: (N/A)
  State: Ringing (5)
  Rings: 0
   NativeFormat: 8
WriteFormat: 8
 ReadFormat: 8
1st File Descriptor: 366
  Frames in: 0
 Frames out: 1240

With wich codec is the channel working now, with ALAW or with g.729A 
And what is the relevant value "read/write format" or "nativeformat" ?

Thanks for help,

Thomas.

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[Asterisk-Users] chan_capi

2003-11-06 Thread Nick Knight

Hello All,

 

I have been using chan_capi with eicon diva server 4 Bri for a few
months now and been very happy with the results. I have now got a small
problem. I needed to upgrade - get more lines in - during an office
move. So we moved office and landed with 3 ISDN2e lines.

 

I can place calls perfectly. Problem is Asterisk is only receiving calls
on 1 channel - so effectively I am only getting 1 out of 3 calls (BT ran
some protocol tests on the line). The ISDN lines are setup to all share
the same number.

 

I have adjusted capi.conf from

 

;

; CAPI config

;

; CAPI config

;

;

[general]

nationalprefix=0

internationalprefix=00

rxgain=0.8

txgain=0.8

 

[interfaces]

 

;isdnmode=ptp

;

; CAPI config

;

;

[general]

nationalprefix=0

internationalprefix=00

rxgain=0.8

txgain=0.8

 

[interfaces]

 

;isdnmode=ptp

msn=870582

incomingmsn=870582

;incomingmsn=870582,870583,870878,871310,871743,871821,872215,872355

;incomingmsn=*

controller=2

context=capiIN

devices=2

 

To:

 

 

 

; CAPI config

;

;

[general]

nationalprefix=0

internationalprefix=00

rxgain=0.8

txgain=0.8

 

[interfaces]

 

;isdnmode=ptp

;

; CAPI config

;

;

[general]

nationalprefix=0

internationalprefix=00

rxgain=0.8

txgain=0.8

 

[interfaces]

 

;isdnmode=ptp

msn=870582

incomingmsn=870582

;incomingmsn=870582,870583,870878,871310,871743,871821,872215,872355

;incomingmsn=*

controller=1,2,3

context=capiIN

devices=6

 

the asterisk startup script is:

;

#!/bin/bash

case "$1" in

stop)

killall asterisk

rmmod capi

rmmod divacapi

rmmod divas

 

;;

 

start)

modprobe divas

modprobe divacapi

modprobe capi

 

divactrl load -c 1 -f ETSI -u -t 0

 

asterisk

;;

 

I have tried various computations of the config - but don't really
understand it - can someone help please

 

Regards

 

Nick 

 

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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

On Wed, 2003-11-05 at 20:47, Steven Critchfield wrote:

> On a PSTN connection though, you get the problem of physical interfaces.

Yes, I have often wondered about this - if we have a single RJ45
connector from the PRI to the Digium Wildcard, how can we deal with the
failure of the main * box without needing to manually move the cable to
the other machine?

> While it was recently mentioned that there is a device for T1 interfaces
> to fail over in the case of alarm, and this could allow a new machine to
> pick up and deal with calls from the PSTN. 

OK.. so you're talking about a 'PRI switch' that only passes data
through to one of the PBXs on the other side... I wonder how such a
switch is informed of which PBX to use? Heartbeat via serial/LAN? From
both machines I'd imagine?

> Of course as I think back, The Intertel hardware our sister company 
> was installing didn't have any HA features. 

Just curious - since the HA will be a major plus-point for us - the
Inter-Tel Axxess system we currently use has had fairly appalling
results and regular failures.

Of course, neither our PBX reseller will admit their hardware is faulty,
nor will the phone company admit there is any fault on the PRI.. yet we
still experience downtime and lost business. 

At least with * we can see line errors ourselves.

> I wouldn't consider this a CON so much as a classification of what is
> possible to do. I doubt the hardware the person who started this thread
> has has any HA features built into it right now.

Amen to that, and it's the reason why I believe * to be the right choice
for us simply due to the flexibility it leaves us with.. one enormous
server with full Linux HA - or two decent machines, each with their own
PRIs...

Cheers,
Gavin.

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Re: [Asterisk-Users] archives gsm of asterisk ???

2003-11-06 Thread Olle E. Johansson

Shoval Tom wrote:

Olle.

I've been in the mailing list for a couple of weeks now.
Many threads are answered with links to your wiki.
Cause of the DNS problem I can't get there, no matter what.

Till this is resolved, are you able to provide me (and many others) the
legit IP address for the web server for me to put in my hosts file?
64.65.102.50

...and now the domain is back to only one DNS server...
Strange things are afoot.
/Olle

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Re: [Asterisk-Users] 6.0 image for Cisco 7960's?

2003-11-06 Thread Doug Heckaman

I hear bkw_ (on #asterisk) has it on his phone, and he said intercom 
works...



Doug

John Todd wrote:

Has anyone managed to get their hands on a 6.0 image for their 7960's 
yet?  Or is it still in beta?

Rumor (official rumor, from Cisco) is that it supports paging and 
intercom.  I'm anxious to start working with those features, if 
they've been implemented sanely.  What would be just as nice would be 
NOTIFY messages for pushing XML URL's to the phones, but sadly that 
feature request has gone uncommented-upon by Cisco, so I will assume 
the worst...

JT
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Re: [Asterisk-Users] Manager Server

2003-11-06 Thread CW_ASN - Gus

Yes, is posible.

  - Original Message - 
  From: 
  marin 
  blu 
  To: [EMAIL PROTECTED] 

  Sent: Thursday, November 06, 2003 3:22 
  AM
  Subject: [Asterisk-Users] Manager 
  Server

  Hi,

  Is it possible to control * from the TCP Manager Server in order to 
  support CRM systems ?

  MarinBlu

  Do you Yahoo!?Protect 
  your identity with Yahoo! Mail AddressGuard

Re: [Asterisk-Users] Voicemail2 vs Voicemail

2003-11-06 Thread CW_ASN - Gus

Just replace Voicemail by VoiceMail2 and that's all.
Note that new voicemail.conf is a bit different than old voicemail.conf.

Regards,

Gus

- Original Message - 
From: "Rich Adamson" <[EMAIL PROTECTED]>
To: "Asterisk-a-users-list" <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 4:44 AM
Subject: [Asterisk-Users] Voicemail2 vs Voicemail


> 
> >> Wouldn't that break everybody's dialplans where they would have to
> >> replace all occurrences of Voicemail2 with Voicemail and all
> >> occurrences of Voicemailmain2 with Voicemailmain?
> >
> > No, we would register with both names.
> 
> Is it necessary (with reasonably current cvs) to make any changes in the
> *.conf files to use Voicemail2, or is that happening automatically?
> 
> 
> 
> ___
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RE:[Asterisk-Users] New Phone Review: Clipcomm 101

2003-11-06 Thread Rafael Gonzalez Lomeña

HI,

   be carefull with this phone !

   I am several issues when use this phone with * (and h323 or oh323).

   We report to Korean's manufacturer few week ago a doc with this fails:


 1º. PUSHING THE "HOLD" KEY, we listen to a music on hold ... but we cannot
 make a new call to a different extension or recover another call.

 2º. In Asterisk it is defined the service "*01" for a task ... if we dial
 "*01" at your phone, Asterisk doesn´t recognize the "*01" dialling.

 3º. TRANSFER CALL doesn´t work using "Transfer KEY".

 Instead of transfering the call, it stops the comunication and dial
 to the extension dialed.

 4º. If we use Asterisk "Transfer" functionallity, (dialling "#" followed by
 the extension...),

 instead of a transfer a multi conference takes place ...

 (then if the phone who transfered the call hangs down the every party lose
 the conversation ... it might be Asterisk´s fault, or it might be normal).

 5º.  If we make previous test several times, and a normal call is made
 after, it crash down, showing the message:
 ### Error ! ###
 nix/tlibthrd.cxx
 line:496
 Exit is called

 Then we have to reset the terminal.

 6º. If we make a phone call to a "crashed phone", then your terminal do
also
 "crash" (got frozen), showing the same message.


  At this moment, CLIPCOMM101 is not a good idea.

Rafael.


> - Original Message -
> From: "mattf" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, November 05, 2003 5:57 PM
> Subject: [Asterisk-Users] New Phone Review: Clipcomm 101
>
>
> > Hello,
> >
> > I have received yet another new phone today, the ClipComm 101
> > (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html)
> >
> > I bought it for $165 directly from the Korean Manufacturer(No US
> distributer
> > yet). Here are the features:
> >
> > - Built-in NAT functionality, you can switch from Hub to Nat, great for
> home
> > DSL/Cable users
> > - This includes some limited port forwarding functionality, very
> > cool idea(wish GS had that)
> > - PSTN port included, it can be hooked up to both VOIP and a POTS line
> > - Full duplex Speakerphone
> > - large LCD display
> > - SIP compatible, but I could only place outgoing calls through a proxy,
> not
> > directly through Asterisk, incoming calls from any asterisk extension
went
> > through just fine. I couldn't disable proxy and use a local server for
> some
> > reason, but it has old firmware, so maybe after I upgrade it.
> > - Phonebook and call log
> > - they have a wireless bluetooth version too(I didn't get that one
though)
> > - 5 second reboot
> >
> >
> > Conclusion, very cool phone, wish proxy could be disabled, but this
phone
> is
> > really aimed at the home market. NAT/Router functionality is a great
idea,
> > makes NAT traversal on phones a non-issue for home users. The price is
> good
> > but no US distributor means shipping is steep($30 for mine), take a look
> at
> > the specs on their site, your be impressed what you get for the price.
> >
> > MATT---
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>

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RE: [Asterisk-Users] archives gsm of asterisk ???

Setting it in hosts doesn't do me any good.

Trying to surf to http:// 64.65.102.50 gets me to apache test page.
Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
Get a 404 page doesn't exist.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olle E.
Johansson
Sent: Thursday, November 06, 2003 1:05 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???

Shoval Tom wrote:

> Olle.
> 
> I've been in the mailing list for a couple of weeks now.
> Many threads are answered with links to your wiki.
> 
> Cause of the DNS problem I can't get there, no matter what.
> 
> Till this is resolved, are you able to provide me (and many others) the
> legit IP address for the web server for me to put in my hosts file?
64.65.102.50

...and now the domain is back to only one DNS server...
Strange things are afoot.

/Olle

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Re: [Asterisk-Users] archives gsm of asterisk ???

Shoval Tom wrote:

Setting it in hosts doesn't do me any good.

Trying to surf to http:// 64.65.102.50 gets me to apache test page.
Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
Get a 404 page doesn't exist.
 

Its most likely on a name based virtual server.. edit your hosts file on 
your system and put somthing like..

64.65.102.50  www.voip-info.org

Then in your browser just goto http://www.voip-info.org

Later..

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[Asterisk-Users] MtSQL CDR logging

It would appear that the "uniqueid" field is not being populated in the 
MySQL CDR DB.. Is this an obsolete field or is a bug?

Later..

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[Asterisk-Users] How to control dialout in extensions file

2003-11-06 Thread Jacky Chen

Hi, all

I have builded a pbx server for pstn, sip & h.323 users
but i can't find any example extensions.conf for access 
control when users which call longdistance with pstn,

If anyone have good example, please sharing your experience
Thanks very much


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RE: [Asterisk-Users] archives gsm of asterisk ???

I tried that, it gives me the same error as if I hadn't edited the hosts
file.

"gateway timeout expired"


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Thursday, November 06, 2003 2:14 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???

Shoval Tom wrote:

>Setting it in hosts doesn't do me any good.
>
>Trying to surf to http:// 64.65.102.50 gets me to apache test page.
>Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
>Get a 404 page doesn't exist.
>
>
>  
>

Its most likely on a name based virtual server.. edit your hosts file on 
your system and put somthing like..

64.65.102.50  www.voip-info.org

Then in your browser just goto http://www.voip-info.org

Later..

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Re: [Asterisk-Users] MtSQL CDR logging

On Thu, 2003-11-06 at 11:16, WipeOut wrote:
> It would appear that the "uniqueid" field is not being populated in the 
> MySQL CDR DB.. Is this an obsolete field or is a bug?

I have never looked at this package so I've never read any docs for it,
buy my guess is you need to set the 'uniqueid' column to have flag
"auto_increment". Then every time a new record is added, no value is set
for this field, because MySQL will allocate a new unique value
automatically.

cheers,
Gavin.

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Re: [Asterisk-Users] How to control dialout in extensions file

2003-11-06 Thread CW_ASN - Gus

You could use DISA app.

exten => 2101,1,DISA,/opt/pass.txt|default

Where:
/opt/pass.txt is a plain text file with password list.
default is a destination context.

Anyway, please do 'show application disa' from CLI.

Hope this helps,

Gus

- Original Message - 
From: "Jacky Chen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 8:18 AM
Subject: [Asterisk-Users] How to control dialout in extensions file


> Hi, all
> 
> I have builded a pbx server for pstn, sip & h.323 users
> but i can't find any example extensions.conf for access 
> control when users which call longdistance with pstn,
> 
> If anyone have good example, please sharing your experience
> Thanks very much
> 
> 
> ___
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[Asterisk-Users] Festiavl Access rejected

2003-11-06 Thread Bartosz Jozwiak




 
Hello,
 
I am trying to make work Festival with 
Asterisk.
Festival is up and running but when i connect to 
Festival with Asterisk I got following:
 
server    Thu Nov  6 08:37:01 
2003 : Festival server started on port 1314client(1) Thu Nov  6 
08:37:07 2003 : rejected from support not in access list
 
My hosts file looks like that:
 
127.0.0.1        
    localhost
66.178.37.64   support    
support.cq-link.sr
 
Can somebody tell me why I get all the time: 
client(1) Thu Nov  6 08:37:07 2003 : rejected from support not in access 
list
 
Thx
--Bart

RE: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Philipp von Klitzing

Hi!

> Yeah, there was talk a while back about the whole user interface thing for
> Comedian Mail and especially how it works compared to other systems, what
> could be done to make migration easier for the users etc. There were some
> good threads, check the archives, but chances are not much has changed.

See what's coming up in voicemail at any time now:
http://bugs.digium.com/bug_view_page.php?bug_id=156

This patch adds several enhanced features to Comedian Mail. These 
features include: 

- Recording options: cancel and call operator 
- End-of-recording options: accept, review, re-record 
- Maintenance options: removal of short and silent messages 
- Advanced options: call back, reply, envelope, outcall 
- Temporary greetings 
- Navigation options: jump to first/last message 
- Access options: log into mailbox during greeting 


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[Asterisk-Users] Asterisk and SIP Proxy on same machine?

2003-11-06 Thread Kerker Staffan

Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?

But how will that work if I register some SIP accounts directly 
from asterisk (like my SIP provider) but then wanna dial outbound
pure SIP calls via my SER... Has anyone got a functional system like
this up and running? SER/Asterisk on the same machine?

rgds,
/Staffan kerker

-Ursprungligt meddelande-
Från: CW_ASN - Gus [mailto:[EMAIL PROTECTED]
Skickat: den 6 november 2003 12:45
Till: [EMAIL PROTECTED]
Ämne: Re: [Asterisk-Users] How to control dialout in extensions file


You could use DISA app.

exten => 2101,1,DISA,/opt/pass.txt|default

Where:
/opt/pass.txt is a plain text file with password list.
default is a destination context.

Anyway, please do 'show application disa' from CLI.

Hope this helps,

Gus

- Original Message - 
From: "Jacky Chen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 8:18 AM
Subject: [Asterisk-Users] How to control dialout in extensions file


> Hi, all
> 
> I have builded a pbx server for pstn, sip & h.323 users
> but i can't find any example extensions.conf for access 
> control when users which call longdistance with pstn,
> 
> If anyone have good example, please sharing your experience
> Thanks very much
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

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[Asterisk-Users] Voicemail RFC

2003-11-06 Thread rnc Info Lists

Earlier today someone posted a RFC number related to voice mail.
Unfortunatly I deleted the message so have lost the number and don't see
it yet in Google.  Can you please resend that to me?
Thanks,
Robert
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Re: [Asterisk-Users] MtSQL CDR logging

Gavin Hamill wrote:

On Thu, 2003-11-06 at 11:16, WipeOut wrote:
 

It would appear that the "uniqueid" field is not being populated in the 
MySQL CDR DB.. Is this an obsolete field or is a bug?
   

I have never looked at this package so I've never read any docs for it,
buy my guess is you need to set the 'uniqueid' column to have flag
"auto_increment". Then every time a new record is added, no value is set
for this field, because MySQL will allocate a new unique value
automatically.
cheers,
Gavin.
 

I would have thought it would use the uniqueid that Asterisk generates 
for each call..

If not I can probably use your way. :)

Later..

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Re: [Asterisk-Users] Asterisk and SIP Proxy on same machine?

Kerker Staffan wrote:

Hi
Is it possible (or recommended) to run both Asterisk and
say SER on the same physical machine? How about port conflicts?
Maybe the easiest way is to change the default SIP port on Asterisk?
But how will that work if I register some SIP accounts directly 
from asterisk (like my SIP provider) but then wanna dial outbound
pure SIP calls via my SER... Has anyone got a functional system like
this up and running? SER/Asterisk on the same machine?

rgds,
/Staffan kerker
 

Firstly please start new threads, dont just reply to an existing thread 
and change the subject line..

A possible solution to your question is to maybe use 2 IP addresses on 
your server, use one for SER and one for Asterisk..

Later..

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[Asterisk-Users] USB handsets/headsets??

Anyone got any pointers on where to find USB handsets or headsets that 
can be used as the audio device on a softphone?

Later..

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RE: [Asterisk-Users] New Phone Review: Clipcomm 101

2003-11-06 Thread mattf

Hello,

I was testing the SIP version and didn't have any crashing issues, what
firmware version were you using? I am using the October 27th, 2003 version.

MATT---




-Original Message-
From: Rafael Gonzalez Lomeña [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 6:03 AM
To: [EMAIL PROTECTED]
Subject: RE:[Asterisk-Users] New Phone Review: Clipcomm 101


HI,

   be carefull with this phone !

   I am several issues when use this phone with * (and h323 or oh323).

   We report to Korean's manufacturer few week ago a doc with this fails:


 1º. PUSHING THE "HOLD" KEY, we listen to a music on hold ... but we cannot
 make a new call to a different extension or recover another call.

 2º. In Asterisk it is defined the service "*01" for a task ... if we dial
 "*01" at your phone, Asterisk doesn´t recognize the "*01" dialling.

 3º. TRANSFER CALL doesn´t work using "Transfer KEY".

 Instead of transfering the call, it stops the comunication and dial
 to the extension dialed.

 4º. If we use Asterisk "Transfer" functionallity, (dialling "#" followed by
 the extension...),

 instead of a transfer a multi conference takes place ...

 (then if the phone who transfered the call hangs down the every party lose
 the conversation ... it might be Asterisk´s fault, or it might be normal).

 5º.  If we make previous test several times, and a normal call is made
 after, it crash down, showing the message:
 ### Error ! ###
 nix/tlibthrd.cxx
 line:496
 Exit is called

 Then we have to reset the terminal.

 6º. If we make a phone call to a "crashed phone", then your terminal do
also
 "crash" (got frozen), showing the same message.


  At this moment, CLIPCOMM101 is not a good idea.

Rafael.


> - Original Message -
> From: "mattf" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, November 05, 2003 5:57 PM
> Subject: [Asterisk-Users] New Phone Review: Clipcomm 101
>
>
> > Hello,
> >
> > I have received yet another new phone today, the ClipComm 101
> > (http://www.clipcomm.co.kr/eng/e_product/e_product_voip_ip_phone.html)
> >
> > I bought it for $165 directly from the Korean Manufacturer(No US
> distributer
> > yet). Here are the features:
> >
> > - Built-in NAT functionality, you can switch from Hub to Nat, great for
> home
> > DSL/Cable users
> > - This includes some limited port forwarding functionality, very
> > cool idea(wish GS had that)
> > - PSTN port included, it can be hooked up to both VOIP and a POTS line
> > - Full duplex Speakerphone
> > - large LCD display
> > - SIP compatible, but I could only place outgoing calls through a proxy,
> not
> > directly through Asterisk, incoming calls from any asterisk extension
went
> > through just fine. I couldn't disable proxy and use a local server for
> some
> > reason, but it has old firmware, so maybe after I upgrade it.
> > - Phonebook and call log
> > - they have a wireless bluetooth version too(I didn't get that one
though)
> > - 5 second reboot
> >
> >
> > Conclusion, very cool phone, wish proxy could be disabled, but this
phone
> is
> > really aimed at the home market. NAT/Router functionality is a great
idea,
> > makes NAT traversal on phones a non-issue for home users. The price is
> good
> > but no US distributor means shipping is steep($30 for mine), take a look
> at
> > the specs on their site, your be impressed what you get for the price.
> >
> > MATT---
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [Asterisk-Users] New IAX software phone (for WIndows platform)

2003-11-06 Thread Carlos Arnt

Hi Dan,
 
It's a great program, 
Just a question,
it's open source right ?
 
Can i see the code ? I'm a c++ programmer too with some time to spend now (On vacation) :)
So can i help ? Did you plan put in the page a source code for people download too ?
 
About the ACTIVEX idea it's great too !!
 
Again nice job !!
 
Carlos.
 
On Thu, 6 Nov 2003 09:31:11 +0200, Dan wrote:> HI, When I first load the Gui, I get to see directory displayed on>> the right hand (bottom) below RX/X/TX etc info.>>> You mean first 12 memories.. Once a number is dialied, this place is used for Volume control>> but it>>> would>> be nice to see the same directory all the time so that dialing>> would be easier.>>> In order to keep the window as small as possible and still have the> full functionality, the form is changed depend on current status.> When you're in a call, you can not dial another number, so this is> the reason that the memories are not available.>>> Did you think about keeping the Dial and Hangup buttons as>> permanent>>> feature>> instead of switching back and forth? Sometimes, I also see a>> delete button which may not be necessary?>>> You must take a closer look at the functionality. The two function> buttons depend (again) on the cuirrent status.> They can display DIAL/DELETE, HANGUP/-, REDIAL/-, REJECT/- When you> start enter a number using keypad, after the first digit you et the> DELETE button too, in order to be able to correct the numbet. Each> click on this button deletes the last digit.> When the number is dialed from the memory, DELETE button delete the> whole number. I cannot imagine something simpler than that...> Anyway, that is my feedback so far. i will try using some more>> functions>>> and>> let you know if I find something strange.> Please check it closer. I still work on the detailed help file> which will be available during the weekend.>>> By the way, I had liked the idea of multi-line phone as a drop->> down action as suggested by someone earlier. Default could be a>> single line with basic features.>>> In this moment, the phone is very close in functionality with a> full featured single line analog phone (with callerid, callwaiting,> callwating callerid, and so on)> I do not see an imediate reason to have more than one line for a> standar phone. Do you?> Keep it up.> I'll do it...>>> Thanks for your feedback and best regards,> Dan>>> ___> Asterisk-Users mailing list>[EMAIL PROTECTED]> http://listsdigium.com/mailman/listinfo/asterisk-users


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Re: [Asterisk-Users] USB handsets/headsets??

On Thu, 2003-11-06 at 12:22, WipeOut wrote:
> Anyone got any pointers on where to find USB handsets or headsets that 
> can be used as the audio device on a softphone?
> 

A quick google for 'usb headset' turned up:

http://www.fuw.edu.pl/~pliszka/linux-USB/#labtec

amongst others :)

nice, since they then just appear as a /dev/dsp OSS audio device...

Cheers,
Gavin


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RE: [Asterisk-Users] USB handsets/headsets??

2003-11-06 Thread tan

We sell some bargain basement versions. Check www.telappliant.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 06 November 2003 12:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] USB handsets/headsets??

Anyone got any pointers on where to find USB handsets or headsets that 
can be used as the audio device on a softphone?

Later..

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Re: [Asterisk-Users] USB handsets/headsets??

Hi,

Try something better.
Use a Bluetooth Headset and an USB Bluetooth dongle with AudioGateway
Profile support.
Then you will have a cordless headset which can be used 10m around the PC.

I have the following combination wich works very nice with my DIAX phone.
- headset Plantronics M1000
- MSI 6967 Bluettth dongle(with the latest MSI drivers).

When a call arrives, the ring is played through PC Speaker (you can use the
soundcard for other purpose: MP3 and so on). I can click on the headset
button cu connect to DIAX, then answer the phone.

What I want to implement now is to put an auto mode in my application who
will force connection with the headset when a new call is arrived (like it
happen in a GSM bluettoth phone) and to hear the ring in the headset too.
If someone can help with this, please send me a mail directly.

A future DIAX version will permit to assign Voice (or even pictures) as
ring, so you will be able to hear/show who's calling and answer the phone
from the headset without even touching the PC.

What do you think?

Thanks and best regards,
Dan

- Original Message - 
From: "WipeOut" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 2:22 PM
Subject: [Asterisk-Users] USB handsets/headsets??


> Anyone got any pointers on where to find USB handsets or headsets that
> can be used as the audio device on a softphone?
>
> Later..
>
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>
>

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Re: [Asterisk-Users] New IAX software phone (for WIndows platform)

Hi Carlos,

>- Original Message - 
>From: Carlos Arnt
>To: [EMAIL PROTECTED]
>Sent: Thursday, November 06, 2003 2:32 PM
>Subject: Re: [Asterisk-Users] New IAX software phone (for WIndows platform)


>Hi Dan,

>It's a great program,
>Just a question,
>it's open source right ?

> Can i see the code ? I'm a c++ programmer too with some time to spend now
(On vacation) :)
> So can i help ? Did you plan put in the page a source code for people
download too ?

Sorry...It is not (yet)..just a free application ( as executable)

> About the ACTIVEX idea it's great too !!
Yup...:-)

> Again nice job !!
Thanks a lot. please test it hardly and send me your comments (bugs,
improvements, etc.)

Best  regards,
Dan

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Re: [Asterisk-Users] RE: *, Fritz!PCI and strange behavior

2003-11-06 Thread Chris Wilson

Hi Patrick and others,

On Tue, 4 Nov 2003, Patrick Lidstone (Personal E-mail) wrote:

> > - Very often, after * runs for a while, it stops recognizing incoming
> >   ISDN calls and refuses to send out ISDN calls.
> 
> I have this. If I try to dial out, I get an "all channels are busy at 
> this time" error, when they are not.

How did you get this error displayed? All I get with CAPI DEBUG is a 
reason code.

> >The funny thing is, 
> >   restarting * or CAPI doesn't work - I have to shutdown both, unplug
> >   and replug the ISDN cable, and then after startup everything works
> >   again. At first, I thought that it might be a bad cable, so I taped
> >   down everything in order to prevent it from moving. This
> > didn't help.
> >   I really do not understand why the thing with the cable is 
> > necessary.
> 
> Straight asterisk restart always clears this condition for me.

We just experienced a problem which looked something like this. Nobody 
could call in or out through the Fritz cards until we restarted Asterisk. 
The CAPI debug output looked like this:

  == CONNECT_IND (PLCI=0x101,DID=510,CID=221,CIP=0x1,CONTROLLER=0x1)
  == DISCONNECT_IND PLCI=0x101 REASON=0
  == CONNECT_IND (PLCI=0x101,DID=510,CID=(null),CIP=0x4,CONTROLLER=0x1)
  == DISCONNECT_IND PLCI=0x101 REASON=0

where CONNECT_IND was immediately followed by DISCONNECT_IND. There was 
none of the normal output:

-- data = 514:901482320681
-- capi request omsn = 514
  == found capi with omsn = 514

or CONNECT_CONF/CONNECT_B3_REQ/DISCONNECT_B3_IND.

Reason code 0 looks very suspicious. /var/log/messages showed:

Nov  6 13:11:42 voip kernel: kcapi: appl 1 ncci 0x10102 up
Nov  6 13:12:19 voip kernel: kcapi: appl 1 ncci 0x10102 down
Nov  6 13:13:35 voip kernel: kcapi: appl 1 ncci 0x10101 up
Nov  6 13:13:41 voip kernel: kcapi: appl 1 ncci 0x10101 down
Nov  6 13:13:49 voip kernel: kcapi: appl 1 ncci 0x10101 up
Nov  6 13:13:58 voip kernel: kcapi: appl 1 ncci 0x10101 down
Nov  6 13:15:26 voip kernel: kcapi: appl 1 ncci 0x10101 up
Nov  6 13:15:56 voip kernel: kcapi: appl 1 ncci 0x10101 down

Does anybody have any ideas?

> > Any light that you can shine on this would be most helpful. OBTW: the 
> > answer "don't use a Fritz" is not applicable here - I'm trying to 
> > assess the feasibility of making a <300$ ISDN SoHo PBX...
> 
> I think the problem may be related to call progress indication from the
> ISDN line. I have UK ISDN2e (packaged as Business Highway - which
> includes what is effectively a telco-owned TA with two analogue ports).
> I have noticed that outgoing channels getting tied up corresponds to
> placing a call which is terminated prematurely (e.g. hangup before
> completing dialing) or dialing a call which can't be completed because
> the dialed PSTN subscriber number is invalid. I've learned to live with
> it - but it would be great to get to the bottom of it.

Sorry, forgot to run "show channels" and "capi info". Will try to remember 
next time.

Cheers, Chris.
-- 
   ___ __ _
 / __// / ,__(_)_  | Chris Wilson -- UNIX Firewall Lead Developer |
/ (_ / ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk |
\ _//_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 |

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Re: [Asterisk-Users] USB handsets/headsets??

Dan wrote:

Hi,

Try something better.
Use a Bluetooth Headset and an USB Bluetooth dongle with AudioGateway
Profile support.
Then you will have a cordless headset which can be used 10m around the PC.
I have the following combination wich works very nice with my DIAX phone.
- headset Plantronics M1000
- MSI 6967 Bluettth dongle(with the latest MSI drivers).
When a call arrives, the ring is played through PC Speaker (you can use the
soundcard for other purpose: MP3 and so on). I can click on the headset
button cu connect to DIAX, then answer the phone.
What I want to implement now is to put an auto mode in my application who
will force connection with the headset when a new call is arrived (like it
happen in a GSM bluettoth phone) and to hear the ring in the headset too.
If someone can help with this, please send me a mail directly.
A future DIAX version will permit to assign Voice (or even pictures) as
ring, so you will be able to hear/show who's calling and answer the phone
from the headset without even touching the PC.
What do you think?

Thanks and best regards,
Dan
 

Hi Dan,

I had looked at that option but it seems that the only USB bluetooth 
adapters that support the headset profile and audio gateway profile are 
the very expesive ones, by the time you have got the headset and the USB 
adapter it works out cheaper to buy a hard phone..

I will take a look at the MSI adapter you spoke of and see what they are 
priced like..

Later..

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Re: [Asterisk-Users] USB handsets/headsets??

2003-11-06 Thread Peer Oliver schmidt

Anyone got any pointers on where to find USB handsets or headsets that 
can be used as the audio device on a softphone?
Have look at http://www.voipvoice.com. A UK company offering
a) USB handset with on/off hook
b) USB handset with dialing pad.
They have an API for Windows, which is needed to utilize the dialing 
pad, afaik.

rgds
pos
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Re: [Asterisk-Users] USB handsets/headsets??

Hi,

> ...
> Hi Dan,
>
> I had looked at that option but it seems that the only USB bluetooth
> adapters that support the headset profile and audio gateway profile are
> the very expesive ones, by the time you have got the headset and the USB
> adapter it works out cheaper to buy a hard phone..

Take care that the specification presented on the web page is not allways
updated.
Check the latest drivers too..
For example, the MSI version is not presented as supporting Audio Gateway
profile, but it does with  the latest software.
The same with the very popular Mitsumi Dongle (just the version WIF C5xxx,
not 4xxx
If you have a 4xx one, then you can change it at Mitsunmi (DE) for free with
a new one (just a difference in fimware, but you cannot upgrade it
yourself).

>
> I will take a look at the MSI adapter you spoke of and see what they are
> priced like..


Check this one... it was less than 40 EURO including VAT
http://www.msi.com.tw/program/products/communication/cmu/pro_cmu_detail.php?UID=326&MODEL=
It works great.
The software is labeled MSI, but it is Widcomm based.

BR,
Dan

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[Asterisk-Users] RE: USB handsets/headsets?? (WipeOut)

2003-11-06 Thread Nick Knight

Hello

 

Headsets we use are Plantronic's which are great, but I am still looking
for a USB handset.

 

Regards

 

Nick

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Re: [Asterisk-Users] Reasons why I shouldn't use Asterisk?

2003-11-06 Thread Ariel Batista

-- Original Message --
From: Tilghman Lesher <[EMAIL PROTECTED]>

>On Wednesday 05 November 2003 18:41, Ariel Batista wrote:
>> >> and the biggest one I feel is a major problem!
>> >>
>> >> 5) Voicemail can not be configured unless you re program it
>> >> yourself. And is not based on any standards!
>> >
>> >I'm curious as to what you find unconfigurable in Voicemail.  I'm
>> >also wondering if you have an RFC for voicemail in mind (for
>> >standardization).
>>
>> What the major problem is folders and how they work!  Also once you
>> get into the folders the prompts will not play to what to do with
>> them. You have to pick advanced options to know what the other keys
>> do!  You can not move around fast and if you press the wrong key it
>> will undelete the message and it puts it in the old messages folder. 
>> Users then complain that there light is still flashing.  Most other
>> voicemail system if you delete the message it moves to the next!  And
>> you can configure it not to have delete folder and old folders.
>
>OK, let me get this straight.  Because the Asterisk voicemail menu is
>fault tolerant and lets you undo a delete, it's therefore unacceptable.
>
>It sounds more like you're having a slight learning curve with getting
>up to speed on a new system.  I'm not sure why you fault Asterisk for
>this, as every system out there is going to have a learning curve.  As
>Steve Underwood pointed out elsewhere in this thread, there are no
>standards for voicemail applications, so every system is going to be
>different.

I would agree with you that there is a learning curve with the systems.  And that 
there are no standards.  But what we did is replace another voicemail system with 
Asterisks.  And we are planning on changing others the same way.  I do like the 
feature like you said of having the un-delete.  But you should be able to configure 
the features you want!  2nd all the voice mail systems I have used when you delete a 
message it moves on the the next.  Asterisk does not. Then after you hear your new 
messages it just leaves you there. It will not even play the prompt to go into the old 
folders unless you hit the advance option. I like Asterisk and plan on using it.  But 
you have to understand that when you install this to normal users that are used to 
other voice mail system they will call you every day for weeks about this.

>
>Personally, I like the fact that I have to explicitly delete a message
>for it to get deleted and that if I delete a message accidently, I can
>undo that (and more importantly, the non-techies in the office can
>also do that).  This is a feature, not a bug.

Please do not take this the wrong way.  I am trying to get a system that will allow us 
to configure it so we can get it out to more people!  It's far better then the 
nortel's and others in the cost and ability to change. There is a future here.  It 
needs to move out of the geeks (My self included) use and into the mass media!
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Re: [Asterisk-Users] USB handsets/headsets??

2003-11-06 Thread Roy Sigurd Karlsbakk

I have a couple of USB handsets from Clarisys (SP-phone), but I have no
idea of how to interface them...

On Thu, 2003-11-06 at 14:55, Dan wrote:
> Hi,
> 
> > ...
> > Hi Dan,
> >
> > I had looked at that option but it seems that the only USB bluetooth
> > adapters that support the headset profile and audio gateway profile are
> > the very expesive ones, by the time you have got the headset and the USB
> > adapter it works out cheaper to buy a hard phone..
> 
> Take care that the specification presented on the web page is not allways
> updated.
> Check the latest drivers too..
> For example, the MSI version is not presented as supporting Audio Gateway
> profile, but it does with  the latest software.
> The same with the very popular Mitsumi Dongle (just the version WIF C5xxx,
> not 4xxx
> If you have a 4xx one, then you can change it at Mitsunmi (DE) for free with
> a new one (just a difference in fimware, but you cannot upgrade it
> yourself).
> 
> >
> > I will take a look at the MSI adapter you spoke of and see what they are
> > priced like..
> 
> 
> Check this one... it was less than 40 EURO including VAT
> http://www.msi.com.tw/program/products/communication/cmu/pro_cmu_detail.php?UID=326&MODEL=
> It works great.
> The software is labeled MSI, but it is Widcomm based.
> 
> BR,
> Dan
> 
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Re: [Asterisk-Users] USB handsets/headsets??

Roy Sigurd Karlsbakk wrote:

I have a couple of USB handsets from Clarisys (SP-phone), but I have no
idea of how to interface them...
Aren't the Clarisys products a little pricey??

On Thu, 2003-11-06 at 14:55, Dan wrote:

Hi,

...
Hi Dan,
I had looked at that option but it seems that the only USB bluetooth
adapters that support the headset profile and audio gateway profile are
the very expesive ones, by the time you have got the headset and the USB
adapter it works out cheaper to buy a hard phone..

Take care that the specification presented on the web page is not allways
updated.
Check the latest drivers too..
For example, the MSI version is not presented as supporting Audio Gateway
profile, but it does with the latest software.
The same with the very popular Mitsumi Dongle (just the version WIF C5xxx,
not 4xxx
If you have a 4xx one, then you can change it at Mitsunmi (DE) for free with
a new one (just a difference in fimware, but you cannot upgrade it
yourself).

I will take a look at the MSI adapter you spoke of and see what they are
priced like..

Check this one... it was less than 40 EURO including VAT
http://www.msi.com.tw/program/products/communication/cmu/pro_cmu_detail.php?UID=326&MODEL=
It works great.
The software is labeled MSI, but it is Widcomm based.
BR,
Dan
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Re: [Asterisk-Users] Beginners help

2003-11-06 Thread Steve Murphy

Rajiv--

I've just went thru something similar myself, setting up 2 FXO's, and a
4port FXS... 

>From your included config, your FXO context does not tell asterisk to
answer the phone, therefore, it doesn't. You might try something like
this for your "incoming" context in extensions.conf:

exten => s,1,Answer
exten => s,2,Zapateller,nocallerid   ;; try to shake off the
 ;; post-FTC-DoNoCallList Exceptions
exten => s,3,Background,HowdyThere   ;; A short welcome message?
exten => s,3,Dial,Zap/1
exten => s,4,Voicemail(u1)   ;; And, voicemail if no answer?
exten => s,5,Hangup  ;; Done
exten => s,104,Voicemail(b1) ;; Busy -- Voicemail?
exten => s,105,Hangup;; Done


Or somesuch. Haven't put the stuff in priority 4-105 in the s extension
yet myself, but theoretically it should work Good Luck!

murf



On Wed, 2003-11-05 at 22:57, [EMAIL PROTECTED] wrote:

> Hi,
> 
> I recently purchased two X100Ps and a TMD40B.  I have set everything
> up
> according to the instructions however I can't get asterisk to pick up
> an
> incoming call (It just keeps ringing).  Also, when I pick up an
> internal
> extension I get dead air. I searched the mailing list for 3+ hours --
> What
> am I missing?
> 
> Thanks in advance for your help.
> 
> Here are the installion steps that I used:
> 
> 
> /etc/zaptel.conf added:
> fxsks=5-6
> fxoks=1-4
> 
> 
> /etc/asterisk/zapata.conf added:
> signalling=fxs_ks ; X100P
> group=1
> context=incoming
> channel => 5-6
> 
> signalling=fxo_ks ; TDM40B
> group=2
> context=internal
> channel => 1-4
> 
> 
> 
> /etc/asterisk/extensions.conf added:
> [incoming]
> exten => s,1,Dial,Zap/1
> ;exten => s,1,Dial,Zap/5
> 
> [internal]
> exten => 34,1,Dial,Zap/1
> exten => 823,1,Dial,Zap/2
> exten => 400,1,Dial,Zap/3
> exten => 500,1,Dial,Zap/4
> exten => _9X.,1,Dial,Zap/5/${EXTEN}
> 
> 
> lsmod outputs:
> Module  Size  Used byTainted: P
> wcfxo   7424   0  (unused)
> wcfxs  15808   0  (unused)
> zaptel183072   0  [wcfxo wcfxs]
> ppp_generic15676   0  [zaptel]
> slhc4864   0  [ppp_generic]
> usb-ohci   17064   0  (unused)
> usbcore36416   0  [usb-ohci]
> amd74xx 9380   1
> nvnet  25216   2
> 
> 
> "ztcfg -vv **" outputs:
> 
> Zaptel Configuration
> ==
> 
> 
> Channel map:
> 
> Channel 01: FXO Kewlstart (Default) (Slaves: 01)
> Channel 02: FXO Kewlstart (Default) (Slaves: 02)
> Channel 03: FXO Kewlstart (Default) (Slaves: 03)
> Channel 04: FXO Kewlstart (Default) (Slaves: 04)
> Channel 05: FXS Kewlstart (Default) (Slaves: 05)
> Channel 06: FXS Kewlstart (Default) (Slaves: 06)
> 
> 6 channels configured.
> 



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Re: [Asterisk-Users] USB handsets/headsets??

Hi,

- Original Message - 
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 4:07 PM
Subject: Re: [Asterisk-Users] USB handsets/headsets??


> I have a couple of USB handsets from Clarisys (SP-phone), but I have no
> idea of how to interface them...

You can see them in the available sound devices list?
If yes, you can select them in the Audio configuration, inside DIAX and use
them.

BR,
Dan

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Re: [Asterisk-Users] USB handsets/headsets??

> Anyone got any pointers on where to find USB handsets or headsets that
> can be used as the audio device on a softphone?

I am a fan of the DSP-100 from Plantronics (mono USB) but unfortunately I 
have been unable to get it to work nicely under Linux.  Works _great_ under 
Win32 though; we might be standardizing on these, but I want to check out 
one from LabTec too that supposedly works with Linux.

I'm looking for mono headsets -- the stereo ones are nice but block out too 
much office noise IMO.  In this office that is a bad thing.  :-)

Regards,
Andrew
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RE: [Asterisk-Users] USB handsets/headsets??

You need to install some softphone, you can't interface asterisk to the
headset by itself.
Try x-lite from www.xten.com for windows (or Dan the man's DIAX software -
search the archives) 
Or gnuphone for linux.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Thursday, November 06, 2003 5:23 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] USB handsets/headsets??

Roy Sigurd Karlsbakk wrote:

>I have a couple of USB handsets from Clarisys (SP-phone), but I have no
>idea of how to interface them...
>

Aren't the Clarisys products a little pricey??


>
>On Thu, 2003-11-06 at 14:55, Dan wrote:
>  
>
>>Hi,
>>
>>
>>
>>>...
>>>Hi Dan,
>>>
>>>I had looked at that option but it seems that the only USB bluetooth
>>>adapters that support the headset profile and audio gateway profile are
>>>the very expesive ones, by the time you have got the headset and the USB
>>>adapter it works out cheaper to buy a hard phone..
>>>  
>>>
>>Take care that the specification presented on the web page is not allways
>>updated.
>>Check the latest drivers too..
>>For example, the MSI version is not presented as supporting Audio Gateway
>>profile, but it does with  the latest software.
>>The same with the very popular Mitsumi Dongle (just the version WIF C5xxx,
>>not 4xxx
>>If you have a 4xx one, then you can change it at Mitsunmi (DE) for free
with
>>a new one (just a difference in fimware, but you cannot upgrade it
>>yourself).
>>
>>
>>
>>>I will take a look at the MSI adapter you spoke of and see what they are
>>>priced like..
>>>  
>>>
>>Check this one... it was less than 40 EURO including VAT
>>http://www.msi.com.tw/program/products/communication/cmu/pro_cmu_detail.ph
p?UID=326&MODEL=
>>It works great.
>>The software is labeled MSI, but it is Widcomm based.
>>
>>BR,
>>Dan
>>
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>
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>  
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RE: [Asterisk-Users] IAX/SIP Client

2003-11-06 Thread Asterisk


>DIAX will ve available as an Active X too which can be integrated in a
web
>page, but in a future release.

This is great. How close are you on this Dan? At this time, I cann't
think of a better application for IAX than DIAX. It really opens up IAX
to general public.

Regards,
Ricky



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[Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration - Sanity Check ...

2003-11-06 Thread Vledder, Hans

Hi All,

First of all, please no flames about the footer of this message. It's being
inserted into all outgoing email by the server here without asking me
personally first once I send out email. Since I've been very interested in
setting up an Asterisk PBX + IVR + Voicemail system to support a small
business and a foundation for quite some time now, in order to replace their
current ISDN PBX, I've been following the discussions on this list and have
been reading whatever I could find or ran into about Asterisk.

Although my favorite configuration would consist of an ISDN2 telco
connection on the outside and running multiple ISDN phones from a single
point-to-multipoint nt-mode ISDN controller (CologneChip based) on the
inside, this does not seem to be supported very well at this stage. ISDN bus
power, as described by the PBX4Linux initiative, is not much of an issue for
me personally, because all our ISDN equipment have their own power supply.
Hopefully support for internal ISDN S-busses will become available in time,
as we, like many others in Europe, have made significant investments in ISDN
equipment over the years and like to be able to use it with Asterisk.

For the time being I've decided to hookup an Asterisk box to one to the
internal extension ports of our PBX and run an IVR and Voicemail system on
it. Before I do, I would appreciate more experienced Asterisk users to
review my plans and check whether my ambitions are feasible. The
configuration that I would like to setup has the following characteristics:

a. Pentium 166 (am I pushing it ?) based Asterisk box running Linux (what
flavour ?)
b. Hooked up to ISDN PBX using an ISDN4Linux (chan_modem ?) or CAPI4Linux
(chan_capi ?) compatible ISDN controller.
c. Asterisk accepting external calls from MSN's configured in PBX and
transferred to Asterisk box (trivial !)
d. Custom IVR application (module or AGI based ?) presenting caller with
usual IVR stuff according to our needs and transfering (chan_modem or
chan_capi, which one can do it ?) calls to the phones connected to the PBX.
e. Setup voice mail and make sure PBX transfers calls to it when
appropriate.

Can this be done like this ?

Regards,
Hans Vledder
The Netherlands

P.S. He who comes up with clean internal ISDN bus (point to multi-point)
support for Asterisk, based on CologneChip based equipment receives an 18"
large Dutch cheese in the mail, right after I've wiped away my tears of
happiness !



-- 
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in error please notify us immediately
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Re: [Asterisk-Users] USB handsets/headsets??

2003-11-06 Thread Ariel Batista

-- Original Message --
From: Andrew Kohlsmith <[EMAIL PROTECTED]>
>> Anyone got any pointers on where to find USB handsets or headsets that
>> can be used as the audio device on a softphone?
>
>I am a fan of the DSP-100 from Plantronics (mono USB) but unfortunately I 
>have been unable to get it to work nicely under Linux.  Works _great_ under 
>Win32 though; we might be standardizing on these, but I want to check out 
>one from LabTec too that supposedly works with Linux.

Just an to give you another choice we have started to use for windows the Telex H-841 
and the H-851 USB headsets.  They work with X-lite and the new DIAX.  Good sound and 
simple to configure!  They start at around $ 42.00. (I wish they were lower).

>
>I'm looking for mono headsets -- the stereo ones are nice but block out too 
>much office noise IMO.  In this office that is a bad thing.  :-)
>
>Regards,
>Andrew
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Re: [Asterisk-Users] USB handsets/headsets??

Hi,

- Original Message - 
From: "Shoval Tom" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 3:44 PM
Subject: RE: [Asterisk-Users] USB handsets/headsets??


> You need to install some softphone, you can't interface asterisk to the
> headset by itself.

If they have standard linux sound drivers, they can be used together with
the Asterisk CONSOLE too.

BR,
Dan

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Re: [Asterisk-Users] USB handsets/headsets??

Andrew Kohlsmith wrote:

Anyone got any pointers on where to find USB handsets or headsets that
can be used as the audio device on a softphone?
   

I am a fan of the DSP-100 from Plantronics (mono USB) but unfortunately I 
have been unable to get it to work nicely under Linux.  Works _great_ under 
Win32 though; we might be standardizing on these, but I want to check out 
one from LabTec too that supposedly works with Linux.

I didn't see that LabTac had a USB headset..

Later..

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Re: [Asterisk-Users] New Phone Review: Clipcomm 101

> > Too bad its an ugly phone.

> Brian, someone once said beauty is in the eye of the beholder.
> I've seen worse unless my eye deceives me.

While that's true, this beholder tends to agree with Brian...  It's not 
Fisher-Price like the Grandstreams, but it's definately got a very dated 
look to it.  Early 80's-ish IMO.  :-)

Regards,
Andrew
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Re: [Asterisk-Users] X100P + ADSI

2003-11-06 Thread Jayson Vantuyl

> Is there any reason why this combination shouldn't work?
I've never tested it, but I don't know why it wouldn't.  I have had recent
issues with getting an overly hot signal (although that was a channel bank
problem).  You might try cranking the rxgain or txgain into the negative
and see if that helps.

Also, by "not work" what do you mean?  Is the phone completely missing the
ADSI?

-- 
Jayson Vantuyl
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[Asterisk-Users] MP3Player problem

2003-11-06 Thread Areski

Hi all,


Is there something wrong with MP3Player ??? I always get the message
below when I try to play a MP3 :


 -- Executing MP3Player("SIP/phone1-83f9",
"/var/lib/asterisk/mohmp3/02") in
new stack
WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed
out/errored
out with 0


I already saw some old posts about it but no solutions !
part of an old post :
"I found that executing the new mpg123 with: mpg123 sample-hold.mp3 =
sometimes takes a couple of seconds to start playing. Every subsequent =
command (exactly the same) starts playing immediately. Maybe this causes =
the timeout in *?"


Can anyone give me a direction to solve this problem ?
Thanks in advance,
Areski




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Re: [Asterisk-Users] USB handsets/headsets??

2003-11-06 Thread Roy Sigurd Karlsbakk

see attached lsusb file for usb out.
this is a linux box. not windoze. and I can't use windoze for this.


On Thu, 2003-11-06 at 15:26, Dan wrote:
> Hi,
> 
> - Original Message - 
> From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
> To: "Asterisk Users" <[EMAIL PROTECTED]>
> Sent: Thursday, November 06, 2003 4:07 PM
> Subject: Re: [Asterisk-Users] USB handsets/headsets??
> 
> 
> > I have a couple of USB handsets from Clarisys (SP-phone), but I have no
> > idea of how to interface them...
> 
> You can see them in the available sound devices list?
> If yes, you can select them in the Audio configuration, inside DIAX and use
> them.
> 
> BR,
> Dan
> 
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lsusb.out.gz
Description: GNU Zip compressed data

RE: [Asterisk-Users] Questions from a total beginner

2003-11-06 Thread Asterisk










 

Hello Roger,

 

Did you get response from someone on this? I would
like to hear what others have to say about your postings since I am also
interested in similar system.

 

Ricky

 

 

-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marrs Seven
Sent: Sunday, November 02, 2003 10:20 PM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users]
Questions from a total beginner

 



Hello,





 





I would like to setup an * system
but have no experience with Linux and am just learning about VoIP.  My
programming experience is pretty limited as well, so I may be getting in way
over my head, but I am willing to take the time to figure out how to use
*.  





 





I'd like to use * to create a PBX
that initially would include myself and three or four individuals who are
located in separate locations around the US.  After getting that to
work, I'd also like to set up * at another location to provide a local PBX for
a small office environment (just 3 or 4 extentions), that would be linked with
the * server at my location.  It's my understanding that would
enable communication via the net to and from my location and the city where the
second * server would be located, thus eliminating any long distance charges
for calls between the two locations.   





 





I have two PC's that I
want to network together using Linksys 802.11g gear (WRT54G ap/router
& a WMP54G PCI card in my * server).  My main machine
is an XP.  The one I am planning to use for the *
server has an AMD 500 processor; 64mb ram; and  30+ gb of hard drive
available.  I've downloaded the RH9 iso files to install Linux on the
proposed server. I also have one phone line coming into my home that I
would connect to the * server with a Wildcard X100P.





 





Hopefully I've provided enough
background info that my questions will make sense.





 





1) From what I've read, the
hardware for my proposed * server is adequate.  Is that
correct?  Should I put another stick of 64mb ram in the box?





 





2) Is there anything special I
need to know about installing RH9 to work with * and what type of install is
recommended?  Also, it's my understanding that I'll have to install some
additional drivers to get RH9 to work with the WMP54G PCI card, and
maybe the WRT54G also.  I'm confused on that issue so any clarification
would be appreciated.





 





3) I'd like to set up a VoIP phone
at my location, but don't know what brand to use, nor the factors to consider
in making that selection, so suggestions would be great.





 





4) Do the individuals at the
other locations only need to obtain a VoIP phone and the appropriate sound
card in order to gain access * at my location?  Or is there some
additional hardware/software required on their end of the connection?  I
assume that using the same VoIP phone at each location would be the ideal and I
believe that's something we can do, if recommended.





 





5) My * server will be operating
behind NAT on the broadband router, but from what I've read, that can
work, although SIP phones can have some difficulty with NAT.  Can the
VoIP phone used eliminate any problems with NAT?





 





6) What are the pros and cons
if we were to have the various locations (individuals and eventually
the second * server) communicate over a VPN?





 





Thanks in advance for any and all
assistance.





 





Roger

Re: [Asterisk-Users] a bit frightened, guys

> But isn't it likely that many people call 911 simultaneously in case of
> an emergency? Maybe it's not a corner case.

Not here, anyway...  Small company.  :-)

"Someone else is already calling 911.  If you wish to continue with your 911 
call, please press 1.  Otherwise, hang up and calm down."  :-)

Regards,
Andrew
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Re: [Asterisk-Users] Using Asterisk as a VOIP gateway

2003-11-06 Thread Alejandro Ruiz

Hi,
I've done somthing like that with 2 X100p.
basically you connect the both end to any extension of the pbx (fxs port).
when you dail that extension, the first machine will answer and ask for the
extension number of the other end.
So what you actually have, is a local extension tha works as if you picked
up an extension on the other end.
I hope this help...


- Original Message - 
From: "Shoval Tom" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, November 05, 2003 6:12 PM
Subject: RE: [Asterisk-Users] Using Asterisk as a VOIP gateway


> How is it not economical?
> I already have the PBXs on both sides.
> If I switch to * I'll need to get a channel bank
>
> Am I wrong?
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of hkirrc.patrick
> Sent: Wednesday, November 05, 2003 8:36 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Using Asterisk as a VOIP gateway
>
> yes you can but may not be all that economical though.
> on the other hand, if you can replace or do away with
> at least one of the pbx with * at either end,
> i think you'll be ahead of the game :-)
>
>
> Shoval Tomer wrote:
>
> > Is it possible to use * as a VOIP gateway?
> >
> > Can I connect asterisk to one of the trunks on my current PBX and on
> > the other side of the world connect another * to the trunk of another
> > regular PBX - is it possible to transfer calls from here to there?
> >
> > I guess I'll need one port FXO card for each asterisk, but I can't
> > figure how to configure the thing.
> >
> > I know I'll need to configure the regular PBX to forward certain calls
> > to the lines connected to asterisk (by prefix, or just have everyone
> > dial 8 and get a line)
> >
> > Does this scenario make sense to anyone? Or am I barking up the wrong
> > tree?
> >
> > Shoval Tomer , MCSE
> >
> > IT Manager
> >
> > Softov Advanced System Ltd.
> >
> > Email: [EMAIL PROTECTED] 
> >
> > Mobile : 972-55-229220
> >
>
>
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Re: [Asterisk-Users] asterisk bandwidth management

> i am using iLBC codec and IAX.. how can i view the
> bandwidth utilization for this in linux.

I run RRD to gather bytes transferred from all my switch ports.  You could 
do something similar with it and use ifconfig output or even iptables 
counter output.  Works _very_ well and is a breeze to set up.  There are 
many configuration examples and lots of documentation on using RRD with 
SNMP and Linux, just google for them.

Regards,
Andrew
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Re: [Asterisk-Users] Best or any VoIP provider that works with *?

> Suggestions on a VoIP provider that works with *?

I am _very_ happy with voicepulse.  connect.voicepulse.com.  they don't 
treat you like a newbie, it's all configured and billed online and their 
email support is very fast and friendly.  Rates ain't bad, either.  :-)

Oh yes, and they support IAX2 trunking and ILBC, and you can download the 
current rates and your call history in machine-readable format without 
resorting to screen-scraping.  :-)

> The thought of unlimited nationwide calling is of big interest to me and
> others I am sure and I would like to know how others are handling it on
> their end.

They do not have an unlimited plan that I am aware of.   That's fine though; 
unlimited long distance is not an economically viable way to run a 
business, and I would like voicepulse to hang around for a good long while.

Regards,
Andrew
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Re: [Asterisk-Users] ISDN PBX + IVR + Voicemail Configuration - Sanity Check ...

2003-11-06 Thread Klaus-Peter Junghanns

Hi Hans,

Am Don, 2003-11-06 um 15.58 schrieb Vledder, Hans:
> P.S. He who comes up with clean internal ISDN bus (point to multi-point)
> support for Asterisk, based on CologneChip based equipment receives an 18"
> large Dutch cheese in the mail, right after I've wiped away my tears of
> happiness !
> 

I am currently working on a zaptel driver for the hfc-s pci a based ISDN
cards and the modifications to make libpri work with BRIs. And of course
NT mode will be supported. We will also have a 4 port BRI card available
in mid/late november that works in TE and NT mode (onboard termination).
Check out www.junghanns.net/asterisk/ during the next week.

regards

kapejod
-- 
Klaus-Peter Junghanns

CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email:  [EMAIL PROTECTED]
http://www.junghanns.net/asterisk

P.S. i have no idea where i should put the 18" cheese ;-) if it was 19"
i could bring it into the colo...
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RE: [Asterisk-Users] archives gsm of asterisk ???

Guys, it still not working.

Go here
http://www.checkdns.net/quickcheck.aspx?domain=voip-info.org&detailed=1
And see that it returns errors.

PLEASE help.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Thursday, November 06, 2003 2:14 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???

Shoval Tom wrote:

>Setting it in hosts doesn't do me any good.
>
>Trying to surf to http:// 64.65.102.50 gets me to apache test page.
>Trying to surf to http:// 64.65.102.50/tiki-index.php?page=Asterisk
>Get a 404 page doesn't exist.
>
>
>  
>

Its most likely on a name based virtual server.. edit your hosts file on 
your system and put somthing like..

64.65.102.50  www.voip-info.org

Then in your browser just goto http://www.voip-info.org

Later..

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Re: [Asterisk-Users] Red Alarm

> An E1 can be a long way from the box with the right cable. However many
> people use the wrong cable. Using a LAN cable for an E1 often gives
> errors if the cable is more than just a few metres long. Although the
> plugs look the same, the twisted pairs should be grouped differently in
> an E1 cable, and it really makes a difference. If the drop cable is only
> a couple of metres long, a LAN cable is usually adequate. This is also
> true for T1s.

Actually that's not entirely true.

standard 568A/B wired cable does not split pairs for ethernet or DSX1 
wiring.  The problem is that DSX1 uses pins (1,2),(4,5) and ethernet (1,2),
(3,6)  (parenthesis show pairing).  DSX1 must have the (1,2) and (4,5) 
pairs swapped to match the TX to the RX at each end, whereas normal 
ethernet does not, as the switch is cross-wired.  Using an ethernet 
crossover cable does not help since it is swapping (1,2) and (3,6), not 
(1,2) and (4,5).  

The problem with using CAT5 for long telco runs is that the impedance is 
wrong at the line clock rate (~1MHz).  IIRC the impedance for telco is 
specified at 600 ohms @ 1MHz, whereas for CAT5 the impedance is actually 
specified at around 100MHz, where the ethernet line rate is.  You can get 
away with it so long as the impedance is right, but unless you've got the 
data sheets you're playing guessing games.

Regards,
Andrew
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Re: [Asterisk-Users] IAX/SIP Client

Hi Ricky,

- Original Message - 
From: "Asterisk" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 4:54 PM
Subject: RE: [Asterisk-Users] IAX/SIP Client


>
> >DIAX will ve available as an Active X too which can be integrated in a
> web
> >page, but in a future release.
>
> This is great. How close are you on this Dan? At this time, I cann't
> think of a better application for IAX than DIAX. It really opens up IAX
> to general public.

I must first pass two important stepts:
- IAX2 support (as all further development will be based on this)
- cleaning up as much as possible the bugs from the executable version (for
this I really need the help of all the interested users).

Then, if there is a real request for that, ActiveX version can be next..;-)

Best regards,
Dan

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Re: [Asterisk-Users] USB handsets/headsets??

Hi,

- Original Message - 
From: "Roy Sigurd Karlsbakk" <[EMAIL PROTECTED]>
To: "Asterisk Users" <[EMAIL PROTECTED]>
Sent: Thursday, November 06, 2003 5:26 PM
Subject: Re: [Asterisk-Users] USB handsets/headsets??

> see attached lsusb file for usb out.
> this is a linux box. not windoze. and I can't use windoze for this.

then you can try to use them for the Asterisk Console

Sorry that I cannot help you further...
Best regards,
Dan
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Re: [Asterisk-Users] Anyone using * in a live production environment?

> 5) Attempt to balance the hybrid at the 2-line to 4 line interface.

This is _precisely_ why my rollouts are all strongly recommending using a 
channel bank instead of the cheap X100P/TDM400P cards -- a lot of work has 
been put into the hybrid circuitry to dynamically adjust to the line 
impedance.  I've had no serious issues with the X100P/TDM400P in small 
scale stuff but the echo cancel IMO should be done where it originates -- 
at the hybrid.

Having said that, I do have "echocancel=32" in my zapata.conf for the T100P 
connected directly to an Adit600 FXS channel bank.  I also have an old CAC 
AB1 with 12FXS and 12FXO ports I am going to deploy shortly to test things 
like far-end disconnect and other issues.

> be the only real solution.  Part of the problem arises from the use of
> lower impedance telephone wiring nowdays. The typical characteristic
> impedance of Cat5 twisted pair is about 100 ohms and many line cards are
> optimized for a 600 ohm line. This is made worse if the DC resistance of
> the wiring to the CO switch is relatively low.  I haven't tried this

This is a neat idea; something I have not thought of.  However my ideal PSTN 
termination is digital (PRI) ... something to eliminate the hybrid 
altogether, at least on my end.  :-)   For deployments where I am simply 
providing VOIP to an existing phone system, I am recommending installing a 
T100P and a digital trunk for the existing KSU; again to eliminate the 
hybrid mess, or at least push it off to someone else's problem.  :-)

> 6) Try messing with Tx and Rx gains.

Something I have noticed is that on the Adit600 FXS ports, I have had to set 
its RX attenuation to -7dB!!  (TX to -3dB) If my math is correct, that 
means I am attenuating 85% of my incoming signal!  Is this perhaps what you 
are referring to with the super-low impedance?

Thank you for this super technical and informative post.  This is what 
*-users needs... more tech and less running around in circles with the same 
issues over and over!

Regards,
Andrew
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Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users] Setcontext based on CID...)

2003-11-06 Thread Chris Hirsch





I hate doing metoos but I tried to get ahold of Michael Baird and never
got a responsedoes anybody have the AGI code that Michael used for
his Anti-Ex Girlfriend as described below?

Thanks!
Chris

  is the AGI available?
Uriel

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Michael Baird
Sent: Saturday, September 27, 2003 6:37 AM
To: [EMAIL PROTECTED]
Subject: Re: Anti-Ex Girl Friend logic (was Re: [Asterisk-Users]
Setcontext based on CID...)


I do it through AGI, I send the call to an external perl script, check
the called-from-id against a mysql database, then send the call back to
a context based on a ruleset I use, call-approved/call-not-approved/no
digits received. Each context having a different voice message, so that
the caller will know the problem, it works very well.

Regards
MIKE

  
  
Blatantly stolen from Mark's presentation:

exten => 600/2565551212,1,Congestion
exten => 600,1,Dial(Zap/9,15)
exten => 600,2,Voicemail(u600)
exten => 600,102,Voicemail(b600)

If the Caller*ID matches the ex-girlfriend (2565551212), provide
immediate congestion tone. Otherwise try dialing on Zap/9 for up to 15
seconds. If there is no answer send them to voicemail, preceeded by
´unavailable¡ message. If the interface is busy, send them to voicemail
with a ´busy¡ message.


Jeremy McNamara



Matt McIntyre wrote:



  I was wondering if someone might be able to offer a suggestion to me
about how I might go about dropping a caller into a context specific
to their CID. For example, I would like to be able to dial Asterisk
from a specific number (a mobile phone) and have it drop me into a
context other then the one that normal callers receive that has more
options tailored to things I might want to do. I assume that ´answer¡
can somehow be used to do this but I thought I might ask the experts
and see what they might have to say.

Thanks in advance,

(You guys are great)

Matt

^
! Matt McIntyre (KF4FGZ)
! Certified Novell Administrator
! (336) 334-1134 (Campus telephone)
! (336) 215-7199 (Mobile telephone) <- Please note the change
! (336) 334-1134 (Facsimile)
! E-MAIL: [EMAIL PROTECTED] 
! AIM: MixMANJaVa
! ICQ: 11956085
^

  


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Exceptional Dogs for Exceptional People - Help Out Today!

Re: [Asterisk-Users] Questions from a total beginner

Hi,

>-Original Message-
>From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On >Behalf Of Marrs Seven
>Sent: Sunday, November 02, 2003 10:20 PM
>>To: [EMAIL PROTECTED]
>Subject: [Asterisk-Users] Questions from a total beginner

> ...
>I have two PC's that I want to network together using Linksys 802.11g gear
(WRT54G ap/router & a WMP54G PCI
> card in my * server).  My main machine is an XP.  The one I am planning to
use for the * server has an AMD 500
> processor; 64mb ram; and  30+ gb of hard drive available.  I've downloaded
the RH9 iso files to install Linux on the
> proposed server. I also have one phone line coming into my home that I
would connect to the * server with a Wildcard X100P.

The minimum configuration tried for an * server was an old Compaq Armada
1700 ([EMAIL PROTECTED]), 96MB RAM, just IP connections (3 HW phones and several
soft phones), for test purpose only. The system was loaded at a signifiant
level


> 1) From what I've read, the hardware for my proposed * server is adequate.
Is that correct?  Should I put another stick of 64mb ram in the box?
Allways more memory can help.

> 4) Do the individuals at the other locations only need to obtain a VoIP
phone and the appropriate sound card in order to gain access * at my
> location?  Or is there some additional hardware/software required on their
end of the connection?  I assume that using the same VoIP phone
> at each location would be the ideal and I believe that's something we can
do, if recommended.

The cheaper option is to use a SW phone (SIP or IAX based). You need then
just a sound card and a headset.

> 5) My * server will be operating behind NAT on the broadband router, but
from what I've read, that can work, although
> SIP phones can have some difficulty with NAT.  Can the VoIP phone used
eliminate any problems with NAT?
Use IAX... works like a charm behind NAT .

> 6) What are the pros and cons if we were to have the various locations
(individuals and eventually the second * server) communicate over a VPN?
The reliability of the connection is important. I use frequently DIAX
through a low end VPN connection (Microsoft PPTP) and no problems at all.

Best regards,
Dan

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[Asterisk-Users] CVS compile problem on asterisk

2003-11-06 Thread Adams, Gavin

Hi,

Recently I did a fresh CVS checkout of asterisk and am getting the
following errors on compile:

chan_zap.c: In function `zt_train_ec':
chan_zap.c:1078: `ZT_ECHOTRAIN' undeclared (first use in this function)
chan_zap.c:1078: (Each undeclared identifier is reported only once
chan_zap.c:1078: for each function it appears in.)
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
[EMAIL PROTECTED] asterisk]#

make ; make clean on Zapata, zaptel, and libpri. Currently loaded
modules are older, but I'm remiss to make install these three modules
prior to getting a clean build on asterisk too. Makes it easier to roll
back in the event of troubles.

Anyone else seeing this under RH9, kernel 2.4.20-20.9 (completely
up2date)?




Regards,

--- Gavin Adams
Promisant (Technology) Ltd.
Atlanta, GA 

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Re: [Asterisk-Users] Appending a # to the dialed number for upstream carrier.

2003-11-06 Thread Steven Critchfield

On Thu, 2003-11-06 at 00:51, Matteo Brancaleoni wrote:
> hi.
> 
> try with (for example)
> ( is a matched 4 digit number)
> 
> exten => _,1,Dial(Zap/1/${EXTEN}#,30,r)
> 
> that should work.
> The idea is to put a # in the dial app.

While what Matteo suggest above is probably the best option, I'll throw
one more out that may be of use later on as your dialplan gets more
complex.

   -- show application Suffix  --

  -= Info about application 'Suffix' =- 

[Synopsis]:
  Append trailing digits

[Description]:
  Suffix(digits): Appends the  digit  string  specified  by  digits to the
channel's associated extension. For example, the number 555 when  suffixed
with '1212' will become 5551212. This app always returns 0, and the PBX will
continue processing at the next priority for the *new* extension.
  So, for example, if priority  3  of  555 is Suffix 1212, the  next  step
executed will be priority 4 of 5551212. If  you  switch  into an  extension
which has no first step, the PBX will treat it as though the user dialed an
invalid extension.

> David Hindmarsh wrote:
> > Hi,
> > 
> > I have a situation where our upstream carrier needs a # after we have sent
> > the dialed number.
> > 
> > Is this possible.
> > 
> > I have checked the Dial app and tried appending it in the exten but this did
> > not work.
> > 
> > Anybody got an idea.
> > 
> > Thanks in Advance
> > Dave
> > 
> > ___
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> 
> 
> ___
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-- 
Steven Critchfield  <[EMAIL PROTECTED]>

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RE: [Asterisk-Users] MP3Player problem

2003-11-06 Thread David Gomillion

Make sure you have mpg123 installed instead of mpg321... It's in the
archives somewhere... that's what fixed my install.

HTH,
David Gomillion

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Areski
Sent: Thursday, November 06, 2003 9:18 AM
To: Asterisk-Users Mailing-list
Subject: [Asterisk-Users] MP3Player problem

Hi all,


Is there something wrong with MP3Player ??? I always get the message
below when I try to play a MP3 :


 -- Executing MP3Player("SIP/phone1-83f9",
"/var/lib/asterisk/mohmp3/02") in
new stack
WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed
out/errored
out with 0


I already saw some old posts about it but no solutions !
part of an old post :
"I found that executing the new mpg123 with: mpg123 sample-hold.mp3 =
sometimes takes a couple of seconds to start playing. Every subsequent =
command (exactly the same) starts playing immediately. Maybe this causes
=
the timeout in *?"


Can anyone give me a direction to solve this problem ?
Thanks in advance,
Areski




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RE: [Asterisk-Users] H.323 and G729: Another sad tale

2003-11-06 Thread G Lin


Dear all,

I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile
on asterisk/channels/h323.

I also donwload pwlib and openh323 from nufone.net/downloads, and did
following things:

1. /pwlib, make clean, make both
2. /openh323, make clean, make opt
3. /asteriks/channels/h323, make clean, make install, and it is got error
about no chan_h323.o  exists. and the make install is failed.

any one can help on this.

Thanks,
George Lin

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Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk Phone System?

2003-11-06 Thread Stephen R. Besch


Here's a cost analysis, rather crude and inspecific, of using Asterisk
to implement a phone system. I'm really quite naive and new to all this,
so I'd appreciate any corrections, tips, pointers, etc, from those in
the community, who have far greater experience, knowledge, skill, etc.
than I. Am I forgetting something important? Am I way off in my
estimates?
For what it's worth, here's what we actually paid for our system:

   1) CPU: Salvaged from spare parts, estimated cost to purchase:   $500
 Asus A7V133, 900MHz, 256MB,60GB
   2) Two Nics 
 $50
   3) Digium T100P  
   $495
   4) 20 GS Budgetone @$65 
  $1300
   5) Adtran TSU600 Channel Bank (e-bay + patience)   
  $99
   6)  2 Dual FXO plugins for TSU  @$100  (used)
 $200
   7)   1 Dual FXS Plug in for TSU @$100   (used)   
  $100
   8)  Asterisk and Linux 
$0
   9) APC Smart-UPS 700 (used)
 $60

Total:  
  $2804$140/Phone (including phones)

$75/phone (not 
including phones)

Obviously, this is a bit unrealistic for many of you, since there is no 
cost for the networking infrastructure and wiring.  This already existed 
on our site and in fact is provided by our University.  We also have a 
lot of computer and electronics expertise in our lab, which really 
helps.  By the way, the cost was less than 1/2 the estimate for Analog 
PBX hardware from an outside vendor (no installation) and less than 1/3 
of the quote offered for a system with less functionality from the 
University - and we would have had no control over the system and would 
have been charged an addition $5/month per voice mailbox, even if though 
we had to buy all the hardware! Finally, and this is a real plug for *, 
one of our scientists splits his time between Buffalo and Nova Scotia. I 
was able to place an extension at his Nova Scotia office for exactly "0" 
extra cost!  Our University IT department wasn't even able to quote us 
on that functionality.

Stephen R. Besch

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Re: [Asterisk-Users] archives gsm of asterisk ???