date:20040402

I downloaded the app and for the most part have it going.

I have not yet managed to get it to accept the password in the flash 
widget that appears as if it wants to accept it.

I wonder about browser-related problems in that respect: I'm running 
fairly recent Mozilla.

I have also hacked the thing to watch my IAX phones and incoming lines. 
. I need to test a bit and will post my changes.

Thanks.

B.
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Re: AW: [Asterisk-Users] checkout ztdummy

2004-04-02 Thread Glen Gray

On Fri, 2004-04-02 at 10:43, ePyron Felix Deierlein wrote:
 I have tried to follow, but I did not know, wich modul I had to check out..
 
Checkout the Zaptel CVS module. Edit the Makefile in the Zaptel dir to
uncomment the ztdummy source.

-- 
Glen Gray [EMAIL PROTECTED] 17 Dame Court
Senior Software EngineerDublin 2, Ireland
Lincor Solutions Ltd.  Ph: +353 (0) 1 6746413


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[Asterisk-Users] UNSUBSCRIBE

2004-04-02 Thread Abraham Lincoln

 On Fri, 2004-04-02 at 10:43, ePyron Felix Deierlein wrote:
 I have tried to follow, but I did not know, wich modul I had to check
 out..

 Checkout the Zaptel CVS module. Edit the Makefile in the Zaptel dir to
 uncomment the ztdummy source.

 --
 Glen Gray [EMAIL PROTECTED] 17 Dame Court
 Senior Software EngineerDublin 2, Ireland
 Lincor Solutions Ltd.  Ph: +353 (0) 1 6746413


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[Asterisk-Users] SIP call troubleshooting

2004-04-02 Thread Marko Rakar


Can someone help me what went wrong with this call?

This call was initiated from dev/ttyI0 device on my asterisk server to
mediatrix unit. Mediatrix unit user received the call and call started.
I can hear them OK but they can not hear me correctly (cut-off sound,
noise). Call was finally hunged up.

Can anyone point out if there was something wrong?



-*CLI sip debug
SIP Debugging Enabled
Asterisk Ready.
We're at 192.168.3.6 port 12556
Answering/Requesting with preferred capability 8
Answering/Requesting with preferred capability 4
12 headers, 9 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 02 Apr 2004 12:01:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 182

v=0
o=root 10202 10202 IN IP4 192.168.3.6
s=session
c=IN IP4 192.168.3.6
t=0 0
m=audio 12556 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
 (no NAT) to 192.168.3.211:5060
-*CLI

Sip read:
SIP/2.0 180 Ringing
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED];tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 0


7 headers, 0 lines
-*CLI

Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED];tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 152
Content-Type: application/sdp
Contact: 304 sip:[EMAIL PROTECTED]:5060
Allow: INVITE, ACK, BYE, CANCEL, REFER

v=0
o=MxSIP 0 0 IN IP4 192.168.3.211
s=SIP Call
c=IN IP4 192.168.3.211
t=0 0
m=audio 5004 RTP/AVP 8 0
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

10 headers, 8 lines
Found audio format ALAW
Found audio format UNKN
Found description format PCMA
Found description format PCMU
Capabilities: us - 12, them - 12/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: sip:[EMAIL PROTECTED]:5060
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port
to send to
set_destination: set destination to 192.168.3.211, port 5060
Transmitting:
ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED];tag=acc03844-c7bb79c5
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 192.168.3.211:5060
set_destination: Parsing sip:[EMAIL PROTECTED]:5060 for address/port
to send to
set_destination: set destination to 192.168.3.211, port 5060
Reliably Transmitting:
BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED];tag=acc03844-c7bb79c5
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 192.168.3.211:5060
-*CLI

Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
CSeq: 103 BYE
From: 0 sip:[EMAIL PROTECTED];tag=as1dbb6ad3
To: sip:[EMAIL PROTECTED];tag=acc03844-c7bb79c5
Via: SIP/2.0/UDP 192.168.3.6:5060;branch=z9hG4bK4e82ae48
Content-Length: 0


7 headers, 0 lines
-*CLI



The linuX Files -- The Source is Out There. 

mailto:[EMAIL PROTECTED]
http://printel.hr  
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Re: [Asterisk-Users] CISCO 7940 and directory/services problem

2004-04-02 Thread Fran Boon

Simon Brown wrote:
I have quite successfully set up the Services button to work on the 7940
running SIP.
I have a metric-imperial converter, a foreign exchange rate calculator, a
calendar etc available to users.
The XML is really fussy though. 
Could you share these example applications?

Thanks,
Fran.
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Re: [Asterisk-Users] Asterisk + Cisco 7920 + chan_sccp or chan_skinny

2004-04-02 Thread Andy Powell



Alternatively, put it somewhere where we can all get at  it :D

Andy

*** REPLY SEPARATOR  ***

On 02/04/2004 at 06:52 Raymond McKay wrote:

 I am using one version of their chan_sccp with a 7960, and can vouch for
 its functionality there.  If you strike out finding an up-to-date version
 on the net, I can send you a tarball.


I would appreciate it if you could.  I was able to pull v 0.2 from a
website
listed in the archive but it doesn't seem to have the mods for the 7920
listed in the code yet.  I'm assumning this was something put in later CVS
versions but the CVS server no longer seems to be working for the site.  I
believe it should be small enough to email it to me off the list if you
could.  Send to [EMAIL PROTECTED]

Thanks

Raymond McKay
President
RAYNET Technologies LLC
(860) 833-9720
http://www.raynettech.com


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Re: [Asterisk-Users] Asterisk + Cisco 7920 + chan_sccp or chan_skinny

2004-04-02 Thread Jan Czmok

Vic Cross ([EMAIL PROTECTED]) wrote:
 G'day Raymond,
 
 On Thu, 1 Apr 2004, Raymond McKay wrote:
 
  I have seen a few postings in the past regarding the interop of Asterisk and
  the Cisco 7920 WiFi phone.  To date, I have not seen a definitive method to
  getting the phone working.  Assuming someone has this actually working, can
  that person step up and answer these questions.
 
 I don't have 7920, so fail your first requirement, but I do know that the
 7920 is reported to work with chan_sccp as modified by Lambda Solutions
 (their original mods were specifically to provide support for 7920, but
 added other features as well such as multi line registrations for
 79[46]0).  I remember seeing list messages saying that 7920 was completely
 unsupported by either chan_skinny or the original chan_sccp, and even with 
 Lambda's chan_sccp mods only basic function was available.
 
 Where you would get chan_sccp nowadays though is a mystery, as its last
 known download location seems to have disappeared from the net (well, the
 domain appears to have been appropriated and is being used for something
 else)...
 

i am currently moving the server to another location, so the download
area (e.g. cvs et al) should be available by today evening (finally).
If yuo are curious, just use the old ip (193.25.172.2) instead of
cvs.lambda-solutions.de (see main page on www.lmabda-solutions.de)

--jan



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Re: [Asterisk-Users] Questions

2004-04-02 Thread Vic Cross

G'day Jeremy,

On Fri, 2 Apr 2004, Jeremy Bogan wrote:

 Line 1 is the home line, I want to give my DECT cordless phone system 
 it's own extension and this phone will ring when this line is called. 
 I'd like outgoing calls made from the DECT system only to be made from 
 that first line.

Easy.  The key to getting this to work is the concept of contexts.  
Basically, in extensions.conf you will control which channels can do what,
and what happens when calls originate on different channels, by careful
assignment and inclusion of contexts.  The Asterisk Handbook is probably
the place to start here.

 Line 2 will be the business line which has a feature called Multiple 
 Numbers, where the line is assigned two numbers, the second number 
 rings through with distinctive ring. I want when this line rings to 
 have an auto attendant and then forward the calls based on the choice 
 dialled. When making outgoing calls on this line i'd like to be able to 
 utilise the Line plus the second number on the same line (you dial a 
 prefix).

Let me save you some grief here: out of the box, Asterisk can't cope with
the way Telstra presents distinctive ring.  Check out bug number 1007 (I
think; the bug description is something like Distinctive ring after CID)
in the Mantis at http://bugs.digium.com -- there is a patch there that I
wrote, but it will not apply to current CVS due to changes in that section
of the code (and I have not had time yet to rework it).  If you want more
details on this, let me know off-list.

Other than the distinctive ring issue, Asterisk will do everything you
want in this scenario.  So, you're good to go (unless you want the two
numbers on your business line to be answered by different IVRs; once the
distinctive ring thing is fixed you'll be able to do that).

I have almost exactly this setup in my home-office, including the dialling
out on the alternate number (I've also got automatic long-distance
override codes as well, depending on which channel initiates the call).  
If you get really stuck I can provide config examples.

Hoo-roo,
Vic Cross
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[Asterisk-Users] Best Web Hosting Resource

2004-04-02 Thread Dan Oproiu MarketingTops.com

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Hi All,Best Web Hosting ResourceFor musiconhold.conf file, how can I play the track01.mp3 as default musicwhen it is on hold?Best Web Hosting ResourceBelow conf is it correct?[classes]default = mp3:/var/lib/asterisk/mohmp3/track01.mp3Best Web Hosting ResourcePlease help!Best Web Hosting ResourceThanks,Randalhttp://www.marketingtops.netDo you Yahoo!?
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2004-04-02 Thread Dan Oproiu MarketingTops.com

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Hi All,Office Space Quotes : Get Office Space QuotesFor musiconhold.conf file, how can I play the track01.mp3 as default musicwhen it is on hold?Office Space Quotes : Get Office Space QuotesBelow conf is it correct?Office Space Quotes : Get Office Space Quotes[classes]default = mp3:/var/lib/asterisk/mohmp3/track01.mp3Office Space Quotes : Get Office Space QuotesPlease help!Office Space Quotes : Get Office Space QuotesThanks,Randalhttp://www.blandonnet.ch
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel - keynames

2004-04-02 Thread Diego Ercolani

Il 22:52, giovedì 01 aprile 2004, Nicolas Gudino ha scritto:
 http://sip.house.com.ar/operator

 Best regards,
I've seen that keynames are very strictly.
The problem is that for example CAPI channel, change the name every time with 
a serial number

canal: SIP/GS1
canal: MGCP/[EMAIL PROTECTED]
canal: MGCP/[EMAIL PROTECTED]
canal: CAPI[CONTR1/0515871620]/40
canal: CAPI[CONTR1/0515871620]/40
canal: CAPI[CONTR1/0515871620]/41
canal: CAPI[CONTR1/0515871620]/41
canal: CAPI[CONTR1/0515871620]/41
canal: CAPI[CONTR1/0515871620]/41
canal: MGCP/[EMAIL PROTECTED]
canal: MGCP/[EMAIL PROTECTED]
canal: MGCP/[EMAIL PROTECTED]
canal: CAPI[CONTR1/0515871620]/42
canal: CAPI[CONTR1/0515871620]/42
canal: CAPI[CONTR1/0515871620]/43
canal: CAPI[CONTR1/0515871621]/43
canal: CAPI[CONTR1/0515871621]/43
canal: CAPI[CONTR1/0515871621]/43
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Re: [Asterisk-Users] Modems

2004-04-02 Thread Martin Mielke

Hi Jeremy,

Jeremy Hall wrote:

Actually, the short answer any more is yes, you can use a modem.
 

Cool! that could make my life easier when setting up a demo system to 
sell Asterisk to my bosses... :-)

I know it is better for several reasons to use an actual Digium X100P.
The main reason being that supporting them is a very good thing.  They
are the reason Asterisk exists.  However, I see lots of messages in
various forums wanting something cheap to start out with, and for many
of us, $100 for a card, or $180 for a dev kit just doesn't fit the
budget for a test or hobby system.  Personally I would like to see them
sell a cheaper version, without the support option.  If they sold one
per customer for $50 without the hour of support, I think people would
be more likely to buy one.  I would have, that is for sure.
 

By now I only need a working VoIP-PSTN demo on Asterisk. Buying such 
dedicated telephony cards is the next step.

That being said, you need a specific firmware on the modem, Intel 537 or
MD3200.  

How to find out? For both the built-in modem in my laptop and for the 
external US-Robotics I can't find it on the provided docs...

[ snip ]

Please note that I do not sell any of these cards on eBay, and am not
trying to support any specific seller.  I simply found one the works,
and wanted to help others in low-budget situations out.  I will be happy
to help anyone out that needs it with these cards, but keep in mind that
mine installed with no issues at all, so I don't have any
troubleshooting experience with this card.
 

Could you please provide some help on how to configure Asterisk to use a 
modem for outgoing calls? For outgoing SIP-calls it works fine...

[ snip ]

Thanks and regards,
Martin
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[Asterisk-Users] UNSUBSCRIBE

2004-04-02 Thread Altus Snyman



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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Adams, Gavin

 -Original Message-

 http://sip.house.com.ar/operator

I love these types of applications that show off the capabilities of *.
This was easy to get up and running for my SIP channels, but for some
reason my PRI (ZAP/1 through ZAP/6) aren't showing up. Has anyone else got
this working for SIP - Trunk lines?

 You can also perform some actions. Hang-up channels and Transfers via
 drag and drop.

How hard would it be to disable these functions. We have the need to show
station status to our users, but would like to remove the ability to hang
up other peoples calls.


 The difference with other similar tools is that it displays status in
 real time (no refreshing necessary), and its graphically appealing.

Looking good too! Not being a Flash developer, can the .swf file be
decoded? I'm thinking of changing some colors, making the buttons smaller,
etc. to allow for more channels to be displayed.

Keep up the great work!

Regards,

--- Gavin


smime.p7s
Description: S/MIME cryptographic signature

RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

if you don't give them the pass code they can't hang-up or transfer calls

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Adams, Gavin
Sent: Friday, April 02, 2004 7:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

 -Original Message-

 http://sip.house.com.ar/operator

I love these types of applications that show off the capabilities of *.
This was easy to get up and running for my SIP channels, but for some
reason my PRI (ZAP/1 through ZAP/6) aren't showing up. Has anyone else got
this working for SIP - Trunk lines?

 You can also perform some actions. Hang-up channels and Transfers via
 drag and drop.

How hard would it be to disable these functions. We have the need to show
station status to our users, but would like to remove the ability to hang
up other peoples calls.

 The difference with other similar tools is that it displays status in
 real time (no refreshing necessary), and its graphically appealing.

Looking good too! Not being a Flash developer, can the .swf file be
decoded? I'm thinking of changing some colors, making the buttons smaller,
etc. to allow for more channels to be displayed.

Keep up the great work!

Regards,

--- Gavin
attachment: winmail.dat

RE: [Asterisk-Users] UNSUBSCRIBE

this is not where to send your unsubscribe to
!

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Altus Snyman
Sent: Friday, April 02, 2004 7:20 AM
To: asterisk
Subject: [Asterisk-Users] UNSUBSCRIBE




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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

just type it in it will remain until you restart your browser.  ( it does
not disappear and you do not have to hit enter or anything like that)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch
Sent: Friday, April 02, 2004 3:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

I downloaded the app and for the most part have it going.

I have not yet managed to get it to accept the password in the flash
widget that appears as if it wants to accept it.

I wonder about browser-related problems in that respect: I'm running
fairly recent Mozilla.

I have also hacked the thing to watch my IAX phones and incoming lines.
. I need to test a bit and will post my changes.

Thanks.

B.
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

My apologies to the list members, I sent the mail by mistake to all of you,
while my intention was to send it to Matt Ridell only.

I also made a typo in the naming convention for IAX2, you have to remove the
slash after IAX2.

If you have problems/questions/bug reports with the operator panel, please
send them to me directly! I wont release the .fla source for now, maybe in
the future.

New versions of the application will be posted in
http://sip.house.com.ar/operator , I'm cleaning some bugs in the server and
in the flash applet also. Thanks,

- Original Message - 
From: Nicolas Gudino [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel


 Hi Matt,

 I modify the server to accept IAX2 channels (I think). Can you try it out?
 You have to name them like

 IAX2/[EMAIL PROTECTED]

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[Asterisk-Users] VON show report - Wi Fi Phones

2004-04-02 Thread Stephen Karrington

Title: Message

Is
that wifi phone available? If yes how much and when? I am looking to purchase a
large quantity of wifi phones. I have a few questions on making calls with these
phones and how the accounting of the calls would go. Thanks.

Sincerely,Stephen KarringtonDreamtime.net Inc.http://www.dreamtime.nethttp://www.emailblaster.usCorporate
Office101 California Street, 22nd FloorSan Francisco, CA
94111-5802Voice - 877-203-9308Fax - 310-943-2606Dreamtime is
your global choice for worldwide communication services, viral marketing
software and direct sales channel automation.

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Todd
TaylorSent: Friday, April 02, 2004 10:48 AMTo:
[EMAIL PROTECTED]Subject: RE: [Asterisk-Users] VON
show report
Well, the best new product that I saw was the IAX (or was that SIP?)
WiFi 2500 hard phone thatMark demo'd for everyone at the Mexican joint
after dinner...Hans, do youhave a pic of that sleek, modern yet dare I
say haute couture look? I think that Digium could take over the world
with that thing...maybe it should be the reference platform for SIPFoundry?
;-)

Cheers..Todd

-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James H.
ThompsonSent: Wednesday, March 31, 2004 1:39 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users] VON show
report
I wrote up report on products that caught my interest at
the VON show going on this week in Santa Clara.

http://www.voip-info.org/wiki-VON+Spring+2004+Report

Jim

James H. Thompson[EMAIL PROTECTED]

Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Eric Wieling

Being able to have more buttons as well as changing the button size
would be useful.

On Fri, 2004-04-02 at 08:04, Nicolas Gudino wrote:
 My apologies to the list members, I sent the mail by mistake to all of you,
 while my intention was to send it to Matt Ridell only.
 
 I also made a typo in the naming convention for IAX2, you have to remove the
 slash after IAX2.
 
 If you have problems/questions/bug reports with the operator panel, please
 send them to me directly! I wont release the .fla source for now, maybe in
 the future.
 
 New versions of the application will be posted in
 http://sip.house.com.ar/operator , I'm cleaning some bugs in the server and
 in the flash applet also. Thanks,
 
 - Original Message - 
 From: Nicolas Gudino [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 
  Hi Matt,
 
  I modify the server to accept IAX2 channels (I think). Can you try it out?
  You have to name them like
 
  IAX2/[EMAIL PROTECTED]
 
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[Asterisk-Users] dtmfmode=inband with G.729

2004-04-02 Thread Jim Rosenberg

It appears Asterisk can handle DTMF inband on only a limited selection of
formats, of which G.729 is not one. The issue appears to be something
involving short data -- whatever that is. (I'm inferring all this from
looking at dsp.c in the vicinity of the error message I was getting, which
pointed to line 1424.)

What *is* short data? Is this really a show-stopper for the G.729 format,
or is it just a case that nobody coded this?

I know that RFC 2833 is really a better way to go (this is for h323, so
there is no option dtmfmode=info ...) but I'm not getting that to work. (I
need to change firmware on my Cisco routers to get them to grok rfc2833.)

-T.i.A., Jim
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Hi Eric,

- Original Message - 
From: Eric Wieling [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 11:17 AM
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

 Being able to have more buttons as well as changing the button size
 would be useful.

What screen resolutions do you use, how many buttons do you need?

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[Asterisk-Users] Voicemail Indication Software

2004-04-02 Thread Christopher Lewis

Does anybody know of any software that can show the status of voicemail 
messages?  Or at least provide a visual indication that I have new voicemail?  
Right now I am using Gnophone and I'm checking manually.

Thanks in advance.
-- 
Christopher Lewis

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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Adams, Gavin

We run at 1600x1200, 96 buttons would be useful. 

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Nicolas Gudino
 Sent: Friday, April 02, 2004 9:26 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 Hi Eric,
 
 - Original Message -
 From: Eric Wieling [EMAIL PROTECTED]
 Sent: Friday, April 02, 2004 11:17 AM
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 
  Being able to have more buttons as well as changing the button size
  would be useful.
 
 What screen resolutions do you use, how many buttons do you need?
 
 
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smime.p7s
Description: S/MIME cryptographic signature

[Asterisk-Users] SIP register and externip

2004-04-02 Thread Simon Brown

I put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD
from behind a NAT
With this entry my PSTN calls have a problem in that the other party cannot
hear me - I can hear them.
It does not matter whether I make the call or the other party does.

Any ideas ?

TIA 

Simon Brown

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RE: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

we also would require more buttons, at least 40, can we get a multipage
view.  right know I run multiple servers on the same page to get the effect
of having 3 pages.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Nicolas
Gudino
Sent: Friday, April 02, 2004 8:26 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel


Hi Eric,

- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
Sent: Friday, April 02, 2004 11:17 AM
Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel


 Being able to have more buttons as well as changing the button size
 would be useful.

What screen resolutions do you use, how many buttons do you need?


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Re: [Asterisk-Users] Is asterisks the best for a simple DTMF response system?

2004-04-02 Thread Bob Klepfer

Bryce Nesbitt (mailing list account) wrote:

I received a recommendation to check out Asterisk, as a platform to host
a simple DTMF response system, something like:
   Setup up VoIP endpoint on Linux/FreeBSD system
   Answer incoming VoIP phone calls
   User enters 100#, perl script plays back foo
   User enters 101#, perl script plays back fum
   User enters 102#, perl script looks up something in
  database, converts to text with festival, speaks it.
100, 101 are built in, no perl needed, 102 may require a short script.

How would one get started, using Asterisks on this project, 
Read. http://www.voip-info.org/wiki-Asterisk   Also the config files 
that come with it.  It takes some study time to absorb.
Or hire a consultant.

and is Asterisks the best option?
It's an excellent option, though pretty underutilized for your application.

Is it really good enough for a high volume (though sub carrier-grade) 
solution?
Yes

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[Asterisk-Users] Gnophone installation problems

2004-04-02 Thread Martin Mielke

Hi all,

I installed all needed RPMs by GnoPhone to be installed without problems 
but when attempting to install GnoPhone itself I get this message:

# rpm -Uvh gnophone-0.2.4-1.i386.rpm
error: Failed dependencies:
   mozilla = 0.9.2 is needed by gnophone-0.2.4-1
   libgtkembedmoz.so is needed by gnophone-0.2.4-1
   libgtksuperwin.so is needed by gnophone-0.2.4-1
I'm using Mozilla 1.7a installed from a tarball. The needed libraries 
are just there:

# locate libgtkembedmoz.so
/usr/local/mozilla/libgtkembedmoz.so
# locate libgtkembedmoz.so
/usr/local/mozilla/libgtkembedmoz.so
# locate libgtksuperwin.so
/usr/local/mozilla/libgtksuperwin.so
and the library path includes them:

# grep mozilla /etc/ld.so.conf
/usr/local/mozilla
I sent an email to the GnoPhone support but some weeks ago but, by the 
time I type this, I still haven't seen a reply...

Any thoughts?

Thanks in advance!

Have a nice weekend!
Martin
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[Asterisk-Users] Re: Grandstream and codec G.711

2004-04-02 Thread Doug Meredith

Mireia Munoz de jesus [EMAIL PROTECTED] wrote:

My gateway accepts G.711, but not my Grandstream 100 series SIP phone

Mine does.  It is termed PCMU and PCMA in the Grandstream setup.

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

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[Asterisk-Users] Unsubscribe

2004-04-02 Thread Dave Tipton


DaveTiptonInfrastructureArchitect817-858-9841VoiceEuless,TXHamRadioCallSign:W3DMT--Thedefinitionofinsanityisdoingthesamethingoverandoverandexpectingdifferentresults.--BenjaminFranklin-Original Message-From: Simon Brown [EMAIL PROTECTED]To: [EMAIL PROTECTED]Sent: Sat, 3 Apr 2004 00:38:15 +1000Subject: [Asterisk-Users] SIP register and externipI put an externip=xxx.xxx.xxx.xxx in my sip.conf so I can register with FWD
from behind a NAT
With this entry my PSTN calls have a problem in that the other party cannot
hear me - I can hear them.
It does not matter whether I make the call or the other party does.

Any ideas ?

TIA 

Simon Brown

-
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Re: [Asterisk-Users] dtmfmode=inband with G.729

2004-04-02 Thread Eric Wieling

It's not asterisk, its the codecs.  Codecs other than ulaw and alaw will
distort continuous tones like DTMF.

On Fri, 2004-04-02 at 08:22, Jim Rosenberg wrote:
 It appears Asterisk can handle DTMF inband on only a limited selection of
 formats, of which G.729 is not one. The issue appears to be something
 involving short data -- whatever that is. (I'm inferring all this from
 looking at dsp.c in the vicinity of the error message I was getting, which
 pointed to line 1424.)
 
 What *is* short data? Is this really a show-stopper for the G.729 format,
 or is it just a case that nobody coded this?
 
 I know that RFC 2833 is really a better way to go (this is for h323, so
 there is no option dtmfmode=info ...) but I'm not getting that to work. (I
 need to change firmware on my Cisco routers to get them to grok rfc2833.)
 
 -T.i.A., Jim
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upgrades can NOT be deducted as a gambling loss.

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[Asterisk-Users] X-Lite - Asterisk: Cannot transmit Audio

2004-04-02 Thread Robert Jackson

Title: Message



I am just an 
Asterisk newbie doing a test install. I am using 2 X-Lite clients and 
haveconfigured them according to the wiki on voip-info. A warning is 
still displayed on the Asterisk server console saying that I should disable 
RFC3389 on the client, even after I changed the Transmit Silence to yes. I 
am able to connect and call the other client, but when I do no audio is being 
transmitted by either client. I have verified that audio can be received 
by calling voice mail and I can hear all of the prompts. But when I leave 
a message it always end up blank. I also verified that audio can be 
recorded via a simple sound recorder. I believe that this is a problem 
with the X-Lite not Asterisk, but I am not sure. Your help is greatly 
appreciated.

Thanks,

Robert

Re: [Asterisk-Users] Gnophone installation problems

2004-04-02 Thread Gavin Hamill

On Friday 02 April 2004 16:01, Martin Mielke wrote:
 Hi all,

 I installed all needed RPMs by GnoPhone to be installed without problems
 but when attempting to install GnoPhone itself I get this message:

 # rpm -Uvh gnophone-0.2.4-1.i386.rpm
 error: Failed dependencies:
 mozilla = 0.9.2 is needed by gnophone-0.2.4-1
 libgtkembedmoz.so is needed by gnophone-0.2.4-1
 libgtksuperwin.so is needed by gnophone-0.2.4-1

 I'm using Mozilla 1.7a installed from a tarball. The needed libraries
 are just there:

You've answered your own question. You installed Mozilla from a tarball. RPM 
therefore doesn't know about it. You need to install a recent Mozilla RPM :)

gdh
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Eric Wieling

My users usually use 800x600 and I would need as many buttons as can fit
on that screen. 8-)  One of my servers currently has 18 Zap channels and
6 IAX2 peers.  I switched my laptop to 600x600 and the bottom row of
buttons is cut partially off. 

Another feature, which would be nice is if you would monitor multiple
Asterisk servers from the same screen.  Not a major issue, but would be
nice.

On Fri, 2004-04-02 at 08:25, Nicolas Gudino wrote:
 Hi Eric,
 
 - Original Message - 
 From: Eric Wieling [EMAIL PROTECTED]
 Sent: Friday, April 02, 2004 11:17 AM
 Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
 
 
  Being able to have more buttons as well as changing the button size
  would be useful.
 
 What screen resolutions do you use, how many buttons do you need?
 
 
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  Eric Wieling * BTEL Consulting * 504-899-1387 x2111
In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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RE: [Asterisk-Users] Modems

2004-04-02 Thread Jeremy Hall



-Original Message-
Hi Jeremy,


Jeremy Hall wrote:

Actually, the short answer any more is yes, you can use a modem.

Cool! that could make my life easier when setting up a demo system to 
sell Asterisk to my bosses... :-)
SNIP

Glad I could help, that is why I posted the message to the list when I
saw the latest no modems reply.

By now I only need a working VoIP-PSTN demo on Asterisk. Buying such 
dedicated telephony cards is the next step.

That was the same situation I was in.  I had my system working with a
couple of different SIP and IAX providers, and wanted to make that next
step.  Now I am just waiting for the IAXY to become available to connect
standard phones into the mix as extensions and maybe get a SIP or IAX
phone or two as well.

That being said, you need a specific firmware on the modem, Intel 537
or
MD3200.  

How to find out? For both the built-in modem in my laptop and for the 
external US-Robotics I can't find it on the provided docs...

I can tell you right now that the external USR modem will not work.  It
will have a 3Com chipset on it.  And unless it is USB, it is a serial
modem and doesn't support voice anyway.  Ironically, what you need is a
WinModem that most of us have tried to avoid for so long.  The modem on
your laptop probably won't work either.  But if Linux recognizes it, you
won't break anything by trying.  Just follow the directions as if you
are using an X100P and see if it works.

What I did was do a search on eBay for Intel MD3200 modem and it
returned several auctions where the seller specifically listed the
firmware.  If you have a bunch of modems you want to look at, mine came
with a white Intel sticker on the main chip.  When I removed that label,
it had MD3200 etched on the chip along with some other numbers.

[ snip ]

Could you please provide some help on how to configure Asterisk to use
a 
modem for outgoing calls? For outgoing SIP-calls it works fine...

Your first step is finding a compatible modem and installing it.  In my
case, I physically installed it in my system, and let Kudzu set it up on
the first boot in.  I then followed the instructions for installing an
X100P, and it worked like a charm.  Contrary to popular belief, I did
not need to modify any source code or any odd settings.  Zttools
recognized it as a Generic or Clone (don't remember which) telephony
card.  When I first went in to zttools, it showed the card in a RED
alarm state.  I plugged in the phone line and it happily went to GREEN.

Once you have a working card, there are several examples out there, some
off of the Unofficial Links page on the Asterisk web page, and others on
the Wiki.

Thanks and regards,
Martin

You're very welcome Martin.  If you have any more questions, please feel
free to ask.

Jeremy

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Re: [Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?

2004-04-02 Thread Andrew Kohlsmith

 mmm... I just wondered, since it's very likely that most people ended up
 deleting it *because* of the subject line. .. so it probably wont help ...
 well it might...

I don't know -- It seems that plain English words are not in spam at all these 
days...  It would have read L AGR3 B*REAs3T5 or something...

Hmmm, I smell a new kind of Bayesian style filter coming up...  :-)

-A.
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Re: [Asterisk-Users] Gnophone installation problems

2004-04-02 Thread Glen Gray

On Fri, 2004-04-02 at 16:01, Martin Mielke wrote:
 Hi all,
 
 I installed all needed RPMs by GnoPhone to be installed without problems 
 but when attempting to install GnoPhone itself I get this message:
 
 # rpm -Uvh gnophone-0.2.4-1.i386.rpm
 error: Failed dependencies:
 mozilla = 0.9.2 is needed by gnophone-0.2.4-1
 libgtkembedmoz.so is needed by gnophone-0.2.4-1
 libgtksuperwin.so is needed by gnophone-0.2.4-1
 
 I'm using Mozilla 1.7a installed from a tarball. The needed libraries 
 are just there:
 
 # locate libgtkembedmoz.so
 /usr/local/mozilla/libgtkembedmoz.so
 

I presume you used prebuilt binary rpms then. 
They will most likely have been linked against /usr/lib/mozilla-1.x/

Try getting the gnophone source rpm rebuilding that with

rpmbuild --rebuild gnophone.src.rpm

-- 
Glen Gray [EMAIL PROTECTED] 17 Dame Court
Senior Software EngineerDublin 2, Ireland
Lincor Solutions Ltd.  Ph: +353 (0) 1 6746413


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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Fran Boon

Nicolas Gudino wrote:
http://sip.house.com.ar/operator
Hi Nicholas,

Agree with the other feedback - looks beautiful, the auto-refreshes are 
exceedingly smooth...definitely vindicates using Flash for client-side :)

I also agree that more buttons would be very useful. (Although some of 
my labels get cut-off as-is, so I'd like a slightly smaller font even 
with current size)
In fact I'll have so many that I think what I really want is the option 
to group them into different folders - ideally the user could even 
create their own folder!

Aside from this, I note that the webpage states See at an glance: SIP 
registration status and reachability
How does this work? I can't see any difference on my system between 
registered  unregistered clients (makes a big difference for SoftPhones).

I'd also like to have an option to disable the 'Talking to' part - in 
some situations this might be undesirable.

Thanks a lot for the contribution - I would urge you to continue further :)

Best Wishes,
Fran.
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Re: [Asterisk-Users] Unsubscribe

2004-04-02 Thread Dave Cotton

On Fri, 2004-04-02 at 16:51, Dave Tipton wrote:
  
 
 Dave Tipton
 Infrastructure Architect
 817-858-9841 Voice
 Euless, TX
 Ham Radio Call Sign: W3DMT
 --
 The definition of insanity is doing the same thing over and over and expecting 
 different results. 
 --Benjamin Franklin

Was that Clueless  TX?
Perhaps Franklin said something about that as well.

READ the below and the bottom of this message please.

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Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] sipura fade to static

2004-04-02 Thread Steve Dolloff

Get an RMA.  I've had a few that did that as well.

Stephen Dolloff
DLS Internet Services
847-854-4799 x256
[EMAIL PROTECTED]


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Christopher J. Wolff
 Sent: Thursday, April 01, 2004 5:50 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] sipura fade to static
 
 Hello,
 
 One of the Sipura 2k's I'm using has a dialtone that occasionally
fades to
 static when the user picks up the line.  Are there any settings that I
can
 check that would affect this?
 
 Regards,
 Christopher
 
 
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Stefano Finetti

Very Nice!

I'm experiencing a bit of troubles in using for some kind of channels.

Actually it shows correctly the status only on ZAP/## channels, while i
can't see anything happening on SIP/ channels neither on IAX2/ channels
(neither with the new .pl you posted).

Regards,
-- 
Stefano Finetti

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Re: [Asterisk-Users] Gnophone installation problems

2004-04-02 Thread Fran Boon

Gavin Hamill wrote:
I'm using Mozilla 1.7a installed from a tarball. The needed libraries
are just there:
You've answered your own question. You installed Mozilla from a tarball. RPM 
therefore doesn't know about it. You need to install a recent Mozilla RPM :)
or use --nodeps

F
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Re: [Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?

2004-04-02 Thread Bob Klepfer

Andrew Kohlsmith wrote:

mmm... I just wondered, since it's very likely that most people ended up
deleting it *because* of the subject line. .. so it probably wont help ...
well it might...
   

I don't know -- It seems that plain English words are not in spam at all these 
days...  It would have read L AGR3 B*REAs3T5 or something..
 

You mean like Best Web Hosting Service or Get Office Space Quotes ? :-)

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[Asterisk-Users] Firefly Client can't receive incoming calls

2004-04-02 Thread Ken DeMaria

I'm having a problem configuring asterisk to send incoming calls to
Firefly.I can make outgoing calls from firefly through asterisk
without any problems at all.  The firefly client does this when it's on
the same IP subnet without a firewall, or from a NAT'd environment.  Can
anyone tell me where I'm going wrong?

Here is output from iax2 show peers:

Name/UsernameHost Mask Port  Status
ken-1202/ken-12  63.140.250.98   (D)  255.255.255.255  53618 UNREACHABLE


Here are what I think are the relevant parts of my config:

iax.conf


[ken-1202]
type=friend
host=dynamic
username=ken-1202
secret=XXX
mailbox=1202
permit=0.0.0.0/0.0.0.0
context=default
qualify=1000


extensions.conf

[iax]
exten = 1202,1,Dial(IAX2/[EMAIL PROTECTED],20)
exten = 1202,2,Voicemail,u${EXTEN}
exten = 1202,102,Voicemail,b${EXTEN}

-- 
Ken DeMaria
[EMAIL PROTECTED]




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RE: [Asterisk-Users] Voicemail Indication Software

2004-04-02 Thread John Vogel


http://www.voip-info.org/tiki-index.php?page=Asterisk%20gui%20vmail.cgi 

Or did you mean asynchronously?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christopher
Lewis
Sent: Friday, April 02, 2004 6:27 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Voicemail Indication Software

Does anybody know of any software that can show the status of voicemail
messages?  Or at least provide a visual indication that I have new
voicemail?  
Right now I am using Gnophone and I'm checking manually.

Thanks in advance.
--
Christopher Lewis

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[Asterisk-Users] Asterisk and Zapata... which kernels?

2004-04-02 Thread Gary Franczyk

I need to upgrade the kernel of my Redhat 7.3 (2.4.18-3) box because of a
bug.  Does anyone know what kernel(s) can I use with asterisk-0.7.0,
libpri-0.5.0 and zaptel-0.8.0?


Thanks

Gary F.

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Re: [Asterisk-Users] dtmfmode=inband with G.729

2004-04-02 Thread Jim Rosenberg

On Fri, Apr 02, 2004 at 08:52:09AM -0600, Eric Wieling wrote:
 It's not asterisk, its the codecs.  Codecs other than ulaw and alaw will
 distort continuous tones like DTMF.

Welll ...

At work we experience this with Cisco dial-peers over G.729: DTMF is
erratic. But it's *NOT* inoperable. The way Asterisk does this, it
doesn't even *try* to send the data through. I'd sure like that option,
even if it might not register at the other end.

My Cisco-dial-peer-only connection users tell me that they often have to
try a second time, but DTMF does usually work for them eventually.

Might not resgister does beat Refuse to try ...

If you have an actual IVR application where errors matter, then of
course you might decide you wouldn't want the risk of distored DTMF, but for
simple things like picking an extension on a PBX where the consequence
of an error is just a wrong number, why not give it a go?

Anyone who chooses dtmfmode=inband is knowingly choosing an option that
is inherently error-prone.
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[Asterisk-Users] Welltech FXO: initial tests

2004-04-02 Thread Jorge Mendoza

Hi,

After a long way of problems (shipping, customs, etc) finally I got 
Welltech working. Here below my comments.

- The documentation is poor and have errors
- The web configuration is not complete. However is useful for the basic 
configuration parameters. The command line is necessary for modify all 
parameters.
- The software upgrade is easy. Initially the gw came with H323, we 
upgrade to SIP.
- We have tested only one port, it works well, audio quality is good (alaw).
- Outgoing and incoming calls are working ok.
- The Caller ID (from PSTN side) does not work
- Answer supervision (reversal polarity detection) seems to work fine. 
This feature is very important to us, is the first time that we found 
this feature in a analog CO trunk. In a test application where we play a 
voice message to the called user, the message start to play just after 
answer. Tested with wire phone and cell phones.
- Disconnect tone seems reliable (although the default configuration was 
not adjusted).

We have done dozen of test in order to get the gw working. During the 
tests two issues came up, they need further analysis and tests:
- Two times a UDP packages loop between the gw and * saturated the 
bandwidth after a hung up. Rebooting the gw does not stop the loop. Even 
with the gw turn off, * was sending the packages.Only rebooting * turn 
the system normal.
- The gw port stay locked after a hung up. Apparently due to a no 
detection of the disconnect tone (in this case the tests were carried 
out with a PABX without disconnect tone). But the * user (SIP) was hung 
up and it seems that there are not a release timer.

We will continue the tests and test the Welltech technical support as 
well (no required until now).

Jorge

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[Asterisk-Users] Re: Still trying program - phone call

2004-04-02 Thread John Chambers

Andy Powell wrote:
1 Access to the PSTN - this can be done via a single X100P card (plugs into a standard phone line) or one of the sinlge port T1 cards or 4 port TDM410 cards (if you need a shedload of lines). You can also use a VoIP - PSTN gateway or gateway service (such as, but not limited to, NuFone)

If you use the X100P, then as I say, a standard analogue phone line is all you need (you can add upto 3 X100P's iirc without issues).
If you want to use the T1 cards then you need to get your local telco to deliver T1's to your location
Actually, this is very much like one of our plans. Of course, if we are to install
hardware to connect to the PSTN locally, there's little point in using a complex
package like asterisk. I've done enough modem programming, including implementing
SLIP and PPP from scratch, to know how to handle that. But we were hoping to use
the glorious new VOIP approach, so we won't have to have a staff to babysit the
special hardware. The prospect of lower costs for internet calls also gets people's
attention, but supporting a modem bank is a nightmare that we're hoping to avoid.
Personally I think the medical service should just employ more people for home visits, since seeing a person is better than just hearing an automated voice on the phone.. (you may have differing opinions, but I come from a culture of free healthcare (however bad it is at the moment))
Lots of people would agree with you there. But in modern America, this is becoming
less and less feasible for most of the population.
It looks to me like you put 800 with a context of callme in your .call file...
Nope; there's no 800 anything in any file that I edited. I'm assuming that it's
the result of some default calculation, but I don't know yet.
I suggest that you abide by the adage, learn to walk before you run ... You can take a look at my guide at http://www.automated.it/guidetoasterisk.htm (there are others) which may help clear up one or two points of understanding...
Actually, I already had that bookmarked (and the browser's link coloring gives
away the fact that I've actually read some of it. ;-)
In fact, there's a related topic of sensible hospital communication with the
growing number of medical gadgets that come with networking, especially WiFi,
but also other packet-radio schemes. It sure looks like a good idea. But here
in North America, writing code to send a wireless message is an impressively
difficult task. For example, suppose your gadget detects a medical emergence,
and the 802.11b interface shows a signal. Just fire off a UDP packet, right?
Not if it's, for example, a Starbucks access point. First you have to register
for service, which means that your code has to use a web interface to send in
a credit card number. This is a LOT more code than just a socket() and sendto()
call. And it's different for every commercial WiFi supplier.
But that's a different project. Right now, I'm just trying to demo the baby
step of a routine that sends a message to a phone number and stores the reply
into a file. We know the modem bank would work. I'm trying to find a method
that avoids this, and just uses the Net. We think it's possible. Asterisk got
our attention mostly because the intro docs state clearly that you don't need
special hardware to connect to a phone. But being told it's possible isn't
quite enough, I'm trying to learn how to do it.
You could of course pay my air fare to Boston (and back) and hotel costs and I'd gladly help you out in person.. after my time at VON in Boston last year I wouldn't mind visiting again :D
Yeah; wouldn't it be nice to persuade our employers to pay for this? It reminds
me of advice I've often given musical friends: Don't complain about the way that
people think that other musicians from far away are better than the local yokels.
That way, the local audience pays to fly your friends in for parties and jam
sessions, and their local audience pays to fly you there. You want to encourage
this attitude; it's to everyone's advantage.
OTOH, we're talking about software to do remote communications. It's probably far
better if the developers are forced to do their work across the Net. If you want
the software to work at a distance, it's reassuring to know that the developers
know how to work at a distance.
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Re: [Asterisk-Users] LARGE BREASTS Handoff back to * from * via IAX?

2004-04-02 Thread Andrew Kohlsmith

 I don't know -- It seems that plain English words are not in spam at all
  these days...  It would have read L AGR3 B*REAs3T5 or something..

 You mean like Best Web Hosting Service or Get Office Space Quotes ? :-)

I don't get spam like that.. .it's all misspelled or intentionally obfuscated. 

-A.
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[Asterisk-Users] First approach to Asterisk - need help

2004-04-02 Thread Mariano Sokal

Hello,

We are trying to migrate from an old application based on VOS to some
linux based telephony server. We are investigating bayonne and asterisk,
and we still don't know what is the best option for us.

One of the limitations is our old hardware, we have in stock some old
Dialogic boards. Does asterisk work with such boards? The other
important limitation is that the application needs to interact with a MS
SQL Server in order to answer the users the data they are trying to
retrieve from our database.

Do you believe that Asterisk will do the work for us?

Thanks in advance,

Mariano Sokal
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Re: [Asterisk-Users] Welltech FXO: initial tests

2004-04-02 Thread Joseph Tanner

I have a Welltech 3502 (2 FXS ports) and callerid will not work in SIP
mode.  I contacted Welltech support and they informed me that callerid is
only working with the H.323 firmware.  Once I flashed it with the H.323
firmware and figured out how to get it to work with asterisk, callerid did
indeed start working.

Joseph Tanner
[EMAIL PROTECTED]

 Message: 15
 Date: Fri, 02 Apr 2004 11:13:35 -0500
 From: Jorge Mendoza [EMAIL PROTECTED]
 Organization: TCC S.A.
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Welltech FXO: initial tests
 Reply-To: [EMAIL PROTECTED]

 Hi,

 After a long way of problems (shipping, customs, etc) finally I got
 Welltech working. Here below my comments.

 - The documentation is poor and have errors
 - The web configuration is not complete. However is useful for the basic
 configuration parameters. The command line is necessary for modify all
 parameters.
 - The software upgrade is easy. Initially the gw came with H323, we
 upgrade to SIP.
 - We have tested only one port, it works well, audio quality is good
 (alaw).
 - Outgoing and incoming calls are working ok.
 - The Caller ID (from PSTN side) does not work
 - Answer supervision (reversal polarity detection) seems to work fine.
 This feature is very important to us, is the first time that we found
 this feature in a analog CO trunk. In a test application where we play a
 voice message to the called user, the message start to play just after
 answer. Tested with wire phone and cell phones.
 - Disconnect tone seems reliable (although the default configuration was
 not adjusted).

 We have done dozen of test in order to get the gw working. During the
 tests two issues came up, they need further analysis and tests:
 - Two times a UDP packages loop between the gw and * saturated the
 bandwidth after a hung up. Rebooting the gw does not stop the loop. Even
 with the gw turn off, * was sending the packages.Only rebooting * turn
 the system normal.
 - The gw port stay locked after a hung up. Apparently due to a no
 detection of the disconnect tone (in this case the tests were carried
 out with a PABX without disconnect tone). But the * user (SIP) was hung
 up and it seems that there are not a release timer.

 We will continue the tests and test the Welltech technical support as
 well (no required until now).

 Jorge



 --__--__--

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Re: [Asterisk-Users] Re: Still trying program - phone call

2004-04-02 Thread Andy Powell

On 02/04/2004 at 11:17 John Chambers wrote:

Andy Powell wrote:

 1 Access to the PSTN - this can be done via a single X100P card (plugs
into a standard phone line) or one of the sinlge port T1 cards or 4 port
TDM410 cards (if you need a shedload of lines). You can also use a VoIP -
PSTN gateway or gateway service (such as, but not limited to, NuFone)

 If you use the X100P, then as I say, a standard analogue phone line is
all you need (you can add upto 3 X100P's iirc without issues).
 If you want to use the T1 cards then you need to get your local telco to
deliver T1's to your location

Actually, this is very much like one of our plans.  Of course, if we are
to install
hardware to connect to the PSTN locally, there's little point in using a
complex
package like asterisk.  I've done enough modem programming, including
implementing
SLIP and PPP from scratch, to know how to handle that. But we were hoping
to use
the glorious new VOIP approach, so we won't have to have a staff to
babysit the
special hardware.  The prospect of lower costs for internet calls also
gets people's
attention, but supporting a modem bank is a nightmare that we're hoping to
avoid.

Ok, that's fine except VOIP can bet dodgy - how would you tell for example that the 
audio being delivered wasn't being broken up into unintelligable blips and squeeks 
because of bandwidth issues - particularly over the net?



 Personally I think the medical service should just employ more people
for home visits, since seeing a person is better than just hearing an
automated voice on the phone.. (you may have differing opinions, but I
come from a culture of free healthcare (however bad it is at the moment))

Lots of people would agree with you there. But in modern America, this is
becoming less and less feasible for most of the population.

How true and how sad... :(


 It looks to me like you put  800 with a context of callme in your .call
file...

Nope; there's no 800 anything in any file that I edited. I'm assuming
that it's the result of some default calculation, but I don't know yet.

When in doubt blame aliens..


 You could of course pay my air fare to Boston (and back) and hotel costs
and I'd gladly help you out in person.. after my time at VON in Boston
last year I wouldn't mind visiting again :D

Yeah; wouldn't it be nice to persuade our employers to pay for this?  It
reminds me of advice I've often given musical friends:  Don't complain about the
way that people think that other musicians from far away are better than the local
yokels. That way, the local audience pays to fly your friends in for parties and
jam sessions, and their local audience pays to fly you there.  You want to
encourage this attitude; it's to everyone's advantage.

Yes!



OTOH, we're talking about software to do remote communications. It's
probably far better if the developers are forced to do their work across the Net. If
you want the software to work at a distance, it's reassuring to know that the
developers know how to work at a distance.

Some of us can do it remotely, but I like to visit places - and Boston isn't too 
unlike home (but that's beacuse of it's history :) )...

/me casually increases taxes on tea and runs away...


Andy


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[Asterisk-Users] voicemail

2004-04-02 Thread Chris Clifton

How would one hack the voicemail app to play saved vm messages back in a
'most recent first' fashion ? What source file is this defined in ?

Thanks,
Chris Clifton

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[Asterisk-Users] error with asterisk -vvvvc

2004-04-02 Thread vozip









Hi 



I´m a new user and I do test with my hardware.



I have a
x100p and telephone vozip.



And when I run this command asterisk
c for to test it.

My computer show it warning



[chan_iax.so]
= (Inter Asterisk eXchange)

  == Manager registered action IAX1peers

  == Parsing
'/etc/asterisk/iax1.conf': Not found (No such file or directory)

Apr  2 07:45:12 ERROR[16384]: chan_iax.c:4828 set_config: Unable to load config
iax1.conf

  == Parsing
'/etc/asterisk/iax.conf': Found

  == Using TOS bits 16

  == Registered
channel type 'IAX1' (Inter Asterisk eXchange Drver)

  == Registered
channel type 'IAX' (Inter Asterisk eXchange Drver)

  == IAX Ready and Listening on 0.0.0.0 port 5036

 [chan_sip.so] =
(Session Initiation Protocol (SIP))

  == Parsing
'/etc/asterisk/sip.conf': Found

  == SIP Listening on 0.0.0.0:5060

  == Using TOS bits 0

  == Registered
channel type 'SIP' (Session Initiation Protocol (SIP))

  == Registered
application 'SIPDtmfMode'

 [chan_modem_bestdata.so]
= (BestData (Conexant
V.90 Chipset) VoiceModem Driver)

 [chan_modem_i4l.so] = (ISDN4Linux Emulated
Modem Driver)

 [chan_agent.so]
= (Agent Proxy Channel)

  == Registered
channel type 'Agent' (Call Agent Proxy Channel)

  == Registered
application 'AgentLogin'

  == Registered
application 'AgentCallbackLogin'

  == Parsing
'/etc/asterisk/agents.conf': Found

 [chan_mgcp.so] =
(Media Gateway Control Protocol (MGCP))

  == Parsing
'/etc/asterisk/mgcp.conf': Found

  == MGCP Listening on 0.0.0.0:2427

  == Using TOS bits 0

  == Registered
channel type 'MGCP' (Media Gateway Control Protocol (MGCP))

 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))

Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:6171 load_module: Unable to open IAX timing interface: No such
device

  == Manager registered action IAXpeers

  == Parsing
'/etc/asterisk/iax.conf': Found

Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:5586 set_config: Ignoring port for now

  == Registered
channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))

  == Using TOS bits 16

  == IAX Ready and Listening on 0.0.0.0 port
4569

 [chan_local.so]
= (Local Proxy Channel)

  == Registered
channel type 'Local' (Local Proxy Channel Driver)

 [chan_skinny.so]
= (Skinny Client Control Protocol (Skinny))

  == Parsing
'/etc/asterisk/skinny.conf': Found

  == Skinny listening on 0.0.0.0:2000

  == Registered
channel type 'Skinny' (Skinny Client Control Protocol (Skinny))

 [chan_oss.so] =
(OSS Console
Channel Driver)

Apr  2 07:45:12 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to open /dev/dsp:
No such device

  == No sound card detected -- console channel
will be unavailable

  == Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf

 [chan_phone.so]
= (Linux Telephony API Support)

  == Parsing
'/etc/asterisk/phone.conf': Found

  == Registered
channel type 'Phone' (Standard Linux Telephony API Driver)

 [skipping chan_alsa.so]

 [chan_zap.so] =
(Zapata Telephony w/PRI)

  == Parsing
'/etc/asterisk/zapata.conf': Found

Apr  2 07:45:13 ERROR[16384]: chan_zap.c:7289 setup_zap: Signalling must be specified before any channels
are.

  == Unregistered
channel type 'Tor'

  == Unregistered
channel type 'Zap'

Apr  2 07:45:13 WARNING[16384]: loader.c:312 ast_load_resource: chan_zap.so: load_module failed, returning -1

  == Unregistered
channel type 'Tor'

  == Unregistered
channel type 'Zap'

Apr  2 07:45:13 WARNING[16384]: loader.c:407 load_modules: Loading module chan_zap.so
failed!













Any ideas.?



Cheers..!



vozip

[Asterisk-Users] modprobe wcfxs ------ fail

2004-04-02 Thread vozip











Any ideas..???





[EMAIL PROTECTED]:/etc# modprobe wcfxs

/lib/modules/2.4.24-xfs/misc/wcfxs.o:
init_module: No such device

Hint: insmod
errors can be caused by incorrect module parameters, including invalid IO or
IRQ parameters.

  You may find more information in syslog or the output from dmesg

/lib/modules/2.4.24-xfs/misc/wcfxs.o:
insmod /lib/modules/2.4.24-xfs/misc/wcfxs.o failed

/lib/modules/2.4.24-xfs/misc/wcfxs.o:
insmod wcfxs failed





Cheers..!





vozip

RE: [Asterisk-Users] voicemail

vmail.cgi seems to be written in perl so modifying it should require
knowledge of perl and vi

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Clifton
Sent: Friday, April 02, 2004 10:51 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] voicemail


How would one hack the voicemail app to play saved vm messages back in a
'most recent first' fashion ? What source file is this defined in ?

Thanks,
Chris Clifton

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[Asterisk-Users] ms messenger problems

2004-04-02 Thread Shawn

I have 2 ms messenger clients. I can not talk between them.
It shows them on-line on there PC. But on the contact list it shows them
not online. what can I do?


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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Tony Buser

We're having a problem with transfering calls.  Our channels are not the 
same as the extensions.  We use words instead of numbers.  So our config 
looks like this:

SIP/HRUTTER,1,81101 Hildegard
SIP/JFOLEY-GS,  2,81103 Jerry
Consequently when I drag and drop to transfer a call to Jerry, it fails 
because it tries to transfer to an extension called JFOLEY-GS, but his 
extension is really 81103.  Btw, might want to make the code be a little 
more forgiving, we could only get it to recognize the channels when we 
made the names in all capital letters (SIP/HRUTTER).

I looked through your code to see if I could make some changes, 
unfortunatly I can't speak Italian!  :)

Nicolas Gudino wrote:
http://sip.house.com.ar/operator

Its a server/client combo that displays the status of your Asterisk PBX
in a web browser in real time.
You can also perform some actions. Hang-up channels and Transfers via
drag and drop.
The difference with other similar tools is that it displays status in
real time (no refreshing necessary), and its graphically appealing.
It's a work in progress... so expect some bugs. I appreciate any
feedback you can give me.
Best regards,


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Re: [Asterisk-Users] dtmfmode=inband with G.729

2004-04-02 Thread Steven Critchfield

On Fri, 2004-04-02 at 10:12, Jim Rosenberg wrote:
 On Fri, Apr 02, 2004 at 08:52:09AM -0600, Eric Wieling wrote:
  It's not asterisk, its the codecs.  Codecs other than ulaw and alaw will
  distort continuous tones like DTMF.
 
 Welll ...
 
 At work we experience this with Cisco dial-peers over G.729: DTMF is
 erratic. But it's *NOT* inoperable. The way Asterisk does this, it
 doesn't even *try* to send the data through. I'd sure like that option,
 even if it might not register at the other end.
 
 My Cisco-dial-peer-only connection users tell me that they often have to
 try a second time, but DTMF does usually work for them eventually.
 
 Might not resgister does beat Refuse to try ...

No this is not better. If you allow the tones to be too relaxed you will
trigger them with certain peoples voices. This is known as talk off. We
as a group do not want talk off. In fact we should strive to avoid it as
it is considered a better to not have any talk off. 

 If you have an actual IVR application where errors matter, then of
 course you might decide you wouldn't want the risk of distored DTMF, but for
 simple things like picking an extension on a PBX where the consequence
 of an error is just a wrong number, why not give it a go?
 
 Anyone who chooses dtmfmode=inband is knowingly choosing an option that
 is inherently error-prone.


-- 
Steven Critchfield  [EMAIL PROTECTED]

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Re: [Asterisk-Users] error with asterisk -vvvvc

2004-04-02 Thread Anton Tinchev

vozip wrote:

Hi 

Im a new user and I do test with my hardware.

I have a x100p and telephone vozip.

And when I run this command asterisk c for to test it.
My computer show it warning
[chan_iax.so] = (Inter Asterisk eXchange)
 == Manager registered action IAX1peers
 == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or
directory)
Apr  2 07:45:12 ERROR[16384]: chan_iax.c:4828 set_config: Unable to load
config iax1.conf
 == Parsing '/etc/asterisk/iax.conf': Found
 == Using TOS bits 16
 == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver)
 == Registered channel type 'IAX' (Inter Asterisk eXchange Drver)
 == IAX Ready and Listening on 0.0.0.0 port 5036
[chan_sip.so] = (Session Initiation Protocol (SIP))
 == Parsing '/etc/asterisk/sip.conf': Found
 == SIP Listening on 0.0.0.0:5060
 == Using TOS bits 0
 == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
 == Registered application 'SIPDtmfMode'
[chan_modem_bestdata.so] = (BestData (Conexant V.90 Chipset) VoiceModem
Driver)
[chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver)
[chan_agent.so] = (Agent Proxy Channel)
 == Registered channel type 'Agent' (Call Agent Proxy Channel)
 == Registered application 'AgentLogin'
 == Registered application 'AgentCallbackLogin'
 == Parsing '/etc/asterisk/agents.conf': Found
[chan_mgcp.so] = (Media Gateway Control Protocol (MGCP))
 == Parsing '/etc/asterisk/mgcp.conf': Found
 == MGCP Listening on 0.0.0.0:2427
 == Using TOS bits 0
 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
[chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:6171 load_module: Unable to open
IAX timing interface: No such device
 == Manager registered action IAXpeers
 == Parsing '/etc/asterisk/iax.conf': Found
Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:5586 set_config: Ignoring port
for now
 == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2))
 == Using TOS bits 16
 == IAX Ready and Listening on 0.0.0.0 port 4569
[chan_local.so] = (Local Proxy Channel)
 == Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_skinny.so] = (Skinny Client Control Protocol (Skinny))
 == Parsing '/etc/asterisk/skinny.conf': Found
 == Skinny listening on 0.0.0.0:2000
 == Registered channel type 'Skinny' (Skinny Client Control Protocol
(Skinny))
[chan_oss.so] = (OSS Console Channel Driver)
Apr  2 07:45:12 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to
open /dev/dsp: No such device
 == No sound card detected -- console channel will be unavailable
 == Turn off OSS support by adding 'noload=chan_oss.so' in
/etc/asterisk/modules.conf
[chan_phone.so] = (Linux Telephony API Support)
 == Parsing '/etc/asterisk/phone.conf': Found
 == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
[skipping chan_alsa.so]
[chan_zap.so] = (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
Apr  2 07:45:13 ERROR[16384]: chan_zap.c:7289 setup_zap: Signalling must be
specified before any channels are.
 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
Apr  2 07:45:13 WARNING[16384]: loader.c:312 ast_load_resource: chan_zap.so:
load_module failed, returning -1
 == Unregistered channel type 'Tor'
 == Unregistered channel type 'Zap'
Apr  2 07:45:13 WARNING[16384]: loader.c:407 load_modules: Loading module
chan_zap.so failed!
 

Check zaptel drivers loading





Any ideas.?

Cheers..!

vozip

 

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RE: [Asterisk-Users] voicemail

2004-04-02 Thread Andrew Thompson

Justin Carlson wrote:
 vmail.cgi seems to be written in perl so modifying it should require
 knowledge of perl and vi 
 
The thing is, vmail.cgi isn't the voicemail application.

I've forgotten the password to my * box now so I can't look it up for you.
Look under asterisk/apps for app_voicemail.c or something like that.

-
Andrew Thompson
http://aktzero.com/ 


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Re: [Asterisk-Users] xml output from * ?

Hi,

On Thu, 2004-04-01 at 15:37, John Todd wrote:
 At 9:35 AM -0500 3/31/04, [EMAIL PROTECTED] wrote:
 Hi Yawl,
 
 I took delivery this morning of a used BetaBrite LED
 display sign which I promptly set about playing with.
 Having found a windows app that grabs XML headline
 files from places like Slashdot and CNN as well as
 stocks etc I had an idea.
 
 What if I could get it to display stats from *? Things
 like call volume, queue stats, message waiting info.
 
 Add my voice to the me too chorus, though I don't have the time or 
 skills to write it either.  This would almost certainly be an 
 external application (not in Asterisk) since the manager interface 
 could provide the relevant information.  There are Perl modules for 
 the BetaBrite, I think... dig around.

You can look at the op_server.pl I wrote. It connects to asterisk
manager port, perform some magic and outputs xml to flash clients. It
might give you ideas on how to implement the betabrite interface. Best
regards,

-- 
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.

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RE: [Asterisk-Users] error with asterisk -vvvvc

2004-04-02 Thread vozip

How can do it.???

Where i can find it.?

Cheers.!

Vozip

-Original Message-
From: Anton Tinchev [mailto:[EMAIL PROTECTED] 
Sent: viernes, 02 de abril de 2004 20:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] error with asterisk -c

vozip wrote:

Hi 
 
I´m a new user and I do test with my hardware.
 
I have a x100p and telephone vozip.
 
And when I run this command asterisk c for to test it.
My computer show it warning
 
[chan_iax.so] = (Inter Asterisk eXchange)
  == Manager registered action IAX1peers
  == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or
directory)
Apr  2 07:45:12 ERROR[16384]: chan_iax.c:4828 set_config: Unable to load
config iax1.conf
  == Parsing '/etc/asterisk/iax.conf': Found
  == Using TOS bits 16
  == Registered channel type 'IAX1' (Inter Asterisk eXchange Drver)
  == Registered channel type 'IAX' (Inter Asterisk eXchange Drver)
  == IAX Ready and Listening on 0.0.0.0 port 5036
 [chan_sip.so] = (Session Initiation Protocol (SIP))
  == Parsing '/etc/asterisk/sip.conf': Found
  == SIP Listening on 0.0.0.0:5060
  == Using TOS bits 0
  == Registered channel type 'SIP' (Session Initiation Protocol (SIP))
  == Registered application 'SIPDtmfMode'
 [chan_modem_bestdata.so] = (BestData (Conexant V.90 Chipset) VoiceModem
Driver)
 [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver)
 [chan_agent.so] = (Agent Proxy Channel)
  == Registered channel type 'Agent' (Call Agent Proxy Channel)
  == Registered application 'AgentLogin'
  == Registered application 'AgentCallbackLogin'
  == Parsing '/etc/asterisk/agents.conf': Found
 [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP))
  == Parsing '/etc/asterisk/mgcp.conf': Found
  == MGCP Listening on 0.0.0.0:2427
  == Using TOS bits 0
  == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP))
 [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2))
Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:6171 load_module: Unable to
open
IAX timing interface: No such device
  == Manager registered action IAXpeers
  == Parsing '/etc/asterisk/iax.conf': Found
Apr  2 07:45:12 WARNING[16384]: chan_iax2.c:5586 set_config: Ignoring port
for now
  == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver
2))
  == Using TOS bits 16
  == IAX Ready and Listening on 0.0.0.0 port 4569
 [chan_local.so] = (Local Proxy Channel)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [chan_skinny.so] = (Skinny Client Control Protocol (Skinny))
  == Parsing '/etc/asterisk/skinny.conf': Found
  == Skinny listening on 0.0.0.0:2000
  == Registered channel type 'Skinny' (Skinny Client Control Protocol
(Skinny))
 [chan_oss.so] = (OSS Console Channel Driver)
Apr  2 07:45:12 WARNING[16384]: chan_oss.c:429 soundcard_init: Unable to
open /dev/dsp: No such device
  == No sound card detected -- console channel will be unavailable
  == Turn off OSS support by adding 'noload=chan_oss.so' in
/etc/asterisk/modules.conf
 [chan_phone.so] = (Linux Telephony API Support)
  == Parsing '/etc/asterisk/phone.conf': Found
  == Registered channel type 'Phone' (Standard Linux Telephony API Driver)
 [skipping chan_alsa.so]
 [chan_zap.so] = (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Apr  2 07:45:13 ERROR[16384]: chan_zap.c:7289 setup_zap: Signalling must be
specified before any channels are.
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Apr  2 07:45:13 WARNING[16384]: loader.c:312 ast_load_resource:
chan_zap.so:
load_module failed, returning -1
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Apr  2 07:45:13 WARNING[16384]: loader.c:407 load_modules: Loading module
chan_zap.so failed!
  

Check zaptel drivers loading

 
 
 
 
Any ideas.?
 
Cheers..!
 
vozip

  


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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Justin Carlson wrote:
just type it in it will remain until you restart your browser.  ( it does
not disappear and you do not have to hit enter or anything like that)
I cut and pasted it right from the source code file, but no matter what 
I do, I get the following line in debug:

La clave no coincide --xxx-!

The password that it's reading is in the (null) space between the first 
two hypens, and I converted the real password into x's just in case, 
security-wise.

Maybe it's my version of Mozilla?

B.
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Tony Buser wrote:

I looked through your code to see if I could make some changes, 
unfortunatly I can't speak Italian!  :)

Not that unfortunate; the comments are all in Spanish, not Italian :-)

B.
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[Asterisk-Users] One voicemail - multiple boxes?

I don't want to re-invent the wheel if someone has already hacked a way 
to do this.

One of my customers has a number of stores, and he wants to leave one 
voicemail that would be delivered to all the managers at once.  Each has 
a voicemail account on his server.

I have googled around and looked on the WIKI.  Maybe I'm missing it?

Thanks.

B.
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

Hi Tony,

On Fri, 2004-04-02 at 14:13, Tony Buser wrote:
 We're having a problem with transfering calls.  Our channels are not the 
 same as the extensions.  We use words instead of numbers.  So our config 
 looks like this:
 
 SIP/HRUTTER,1,81101 Hildegard
 SIP/JFOLEY-GS,  2,81103 Jerry
 
 Consequently when I drag and drop to transfer a call to Jerry, it fails 
 because it tries to transfer to an extension called JFOLEY-GS, but his 
 extension is really 81103.  

I will try to take care of that, my asterisk universe is very limited, I
did not think about other naming conventions and uses for the different
types of channels.

 Btw, might want to make the code be a little 
 more forgiving, we could only get it to recognize the channels when we 
 made the names in all capital letters (SIP/HRUTTER).

Version .03 is on the website, case insenstive and more channel types
supported.

 I looked through your code to see if I could make some changes, 
 unfortunatly I can't speak Italian!  :)

Me neither! I speak spanish..LOL.


-- 
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.

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[Asterisk-Users] Newbie Question: ISDN and Capacity Planning

2004-04-02 Thread Chris Travers

Hi all;

I am planning a PBX/Voice mail system for a small business (approx 12 
employees with phones).  They have an inbound ISDN PRI, which is 
probably irrelevant because all inbound calls are routed first to 
receptionists which rarely route the calls on (client is a medical clinic).

Any idea what sort of capacity planning I should be looking at?  Any 
minimum and optimal figures would be good, as my bid will include 
redundant systems.  Any hidden gotchas with ISDN I should be familiar with?

It should be noted that phone use among most of the staff is low to 
average-- calls are rarely routed back to the doctors/nurses but they 
have occasional need to call out.

Best Wishes,
Chris Travers
Metatron Technology Consulting
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[Asterisk-Users] Sorry for the duplicate

2004-04-02 Thread Chris Travers

Hi;

Sorry, I resent a message similar to the parent by mistake.

Best Wishes,
Chris Travers
Metatron Technology Consulting
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[Asterisk-Users] T100P specs

2004-04-02 Thread Ernest W. Lessenger

Does anyone have the physical spec sheet for the T100P from Digium? The one
on the website doesn't have what I need. Things like 3.3 or 5v operation,
uses n IRQ channels, requires 32-bit PCI, must be installed while standing
on one foot and reciting the GPL, etc. Also, if anyone is selling a used
T100P or TE4xxP I'd like to talk...

Thanks,
--Ernest

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Re: [Asterisk-Users] T100P specs

2004-04-02 Thread Andrew Kohlsmith

 Does anyone have the physical spec sheet for the T100P from Digium? The one
 on the website doesn't have what I need. Things like 3.3 or 5v operation,
 uses n IRQ channels, requires 32-bit PCI, must be installed while standing
 on one foot and reciting the GPL, etc. Also, if anyone is selling a used
 T100P or TE4xxP I'd like to talk...

5V, 1 IRQ (INTA), 32-bit PCI slot.  I would recommend using a PCI2.2 compliant 
system although I don't think that's absolutely necessary.

Close on the installation though - It requires a sacrificial SCO exec.  Don't 
have any spares to sell, sorry.

Regards,
Andrew
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Re: [Asterisk-Users] PRI integration with Marconi switch

2004-04-02 Thread Juan J. Sierralta P.

On Thu, 2004-04-01 at 17:32, Scott Stingel wrote:
  Hello-
 
 Has anyone had experience connecting to a Marconi switch (in the UK) using
 E1-PRI connections (TE410P)?  In a new installation, my customer is getting
 yellow alarms on every channel about every 30 seconds.  These alarms clear
 themselves immediately and then re-occur in another 30 seconds, ad
 infinitum.

Are you using last CVS ?
I had the same problem on a TE410 E1 PRI with last CVS, rolling back to
CVS 05/03/2004 solved my problems.

-- 
Juanjo sin .sig

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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel

2004-04-02 Thread Tony Buser

by the way, when I start up op_server.pl I get the following, even 
though everything appears to work ok.

Use of uninitialized value in transliteration (tr///) at ./op_server.pl 
line 67, CONFIG line 35.
Use of uninitialized value in string at ./op_server.pl line 68, CONFIG 
line 35.
Use of uninitialized value in string at ./op_server.pl line 69, CONFIG 
line 35.
Use of uninitialized value in substitution (s///) at ./op_server.pl line 78.
Use of uninitialized value in concatenation (.) or string at 
./op_server.pl line 79.

I looked through your code to see if I could make some changes, 
unfortunatly I can't speak Italian!  :)
Me neither! I speak spanish..LOL.
Woops!  In case you hadn't guessed I don't speak spanish either, sorry.  :)
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Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel