date:20051114

great! i'll try it later tnx!On 11/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Monday 14 November 2005 07:20, Mark Quitoriano wrote: Hi, is there a howto to install 
g.729 codec on asterisk?http://www.digium.com/downloads/ftp/asterisk/g729/README___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] X100P troubles?


  As I recall, the driver for the x100p was called wcfxs (or something 
  like that), and those driver functions were merged into wctdm about a 
  year ago. Now, wcfxs is an alias for wctdm.
 
 What you recall is only partially correct:
 
 In 1.0 wctdm is an alias for wcfxs. Furthermore, wcfxs shows up (e.g: in
 /proc/interrupts) as wctdm.
 
 wcfxs is the driver for the TDM400P cards. It was later renamed
 (rewrriten as?) wctdm. wcfxo is the driver for the X100P cards.

Then that must have switched around after 1.0?

In cvs-head, there is no wcfxs. Its wctdm (and its alias is wcfxs).



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Re: [Asterisk-Users] asterisk 1.0.10?

you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?On 11/14/05, Matt Riddell [EMAIL PROTECTED]
 wrote:Mark Quitoriano wrote: but that's already 1.2? is it advisable to upgrade my current version
 1.0.9 to 1.2 already? any big changes to be done to my current setup to upgrade it to 1.2?If in three or four days you go to upgrade your version of 1.0.9, it will beupgraded to 1.2.So, why not do it now, that way you won't end up creating a dialplan only for
it to not work in a couple of days.
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[Asterisk-Users] Attended transfer and group problem

2005-11-14 Thread Domenico Lanteri


I have a problem with group and attended transfer.
I have tested below example dialplan with asterisk-1.2.0-beta1,
asterisk-1.2.0-rc1 and and astesik-HEAD on 11/14/2005.

I have simple test dialplan like: 

[default]
exten = 210,1,Macro(stdexten,${EXTEN},SIP,test1)
exten = 211,1,Macro(stdexten,${EXTEN},SIP,test2)
exten = 212,1,Macro(stdexten,${EXTEN},SIP,test3)

[macro-stdexten]
exten = s,1,Set(OUTBOUND_GROUP=${ARG1})
exten = s,n,Set(GROUP=${CALLERIDNUM})
exten = s,n,Dial(${ARG2}/${ARG3}||t)

If i dial 211 from 210, 211 answer and make attended trasfer to 212, 212
answer and 211 hangup, cli show: 

Monitor*CLI set verbose 4 
Verbosity is at least 4
-- Executing Macro(SIP/test1-2f4f, stdexten|211|SIP|test2) in new
stack
-- Executing Set(SIP/test1-2f4f, OUTBOUND_GROUP=211) in new stack
-- Executing Set(SIP/test1-2f4f, GROUP=210) in new stack
-- Executing NoOp(SIP/test1-2f4f, EXTEN: -211-  CALLERIDNUM: -210-)
in new stack
-- Executing NoOp(SIP/test1-2f4f, --SIP/test1-2f4f) in
new stack
-- Executing Dial(SIP/test1-2f4f, SIP/test2||t) in new stack
-- Called test2
-- SIP/test2-c1b6 is ringing
-- SIP/test2-c1b6 answered SIP/test1-2f4f
-- Attempting native bridge of SIP/test1-2f4f and SIP/test2-c1b6
-- Attempting native bridge of SIP/test1-2f4f and SIP/test2-c1b6
-- Started music on hold, class 'default', on channel 'SIP/test1-2f4f'
-- Playing 'pbx-transfer' (language 'it')
-- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|212|SIP|test3)
in new stack
-- Executing Set(Local/[EMAIL PROTECTED],2, OUTBOUND_GROUP=212) in
new stack
-- Executing Set(Local/[EMAIL PROTECTED],2, GROUP=211) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2, EXTEN: -212-
CALLERIDNUM: -211-) in new stack
-- Executing NoOp(Local/[EMAIL PROTECTED],2,
--Local/[EMAIL PROTECTED],2) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/test3||t) in new
stack
-- Called test3
-- SIP/test3-61de is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/test3-61de answered Local/[EMAIL PROTECTED],2
Nov 14 12:01:51 NOTICE[6186]: res_features.c:1124
ast_feature_request_and_dial: Don't know what to do about control frame: -1
-- Stopped music on hold on SIP/test1-2f4f
-- Playing 'beep' (language 'en')
  == Spawn extension (macro-stdexten, s, 5) exited non-zero on
'Transfered/SIP/test1-2f4fZOMBIE' in macro 'stdexten'
  == Spawn extension (default, 211, 1) exited non-zero on
'Transfered/SIP/test1-2f4fZOMBIE'
Monitor*CLI group show channels 
ChannelGroup Category
SIP/test1-2f4f 210   (default)   
SIP/test3-61de 212   (default)   
Local/[EMAIL PROTECTED],2   211   (default)   
7 active channels


Channel Local/212 belong to group 211 but the 211 phone is hangup.
Anyone as any idea ?
Thank you
Lanteri Domenico.


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[Asterisk-Users] MYSQL issue in UPDATE..

2005-11-14 Thread Mauro Zanin

Hi Everybody,
I'm trying to execute a MYSQL(UPDATE..) sql
command over a table I have previously red. I get a timeout and no update
happens.
I use  * 1.0.9.
I wonder if MYSQL set of commands allows Update...

Best regards
Mauro Zanin
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Re: [Asterisk-Users] Voicemail file as MP3

where can i find a howto about this?On 11/14/05, Michael Toop [EMAIL PROTECTED] wrote:
Hi,Not natively, but you can run a Bash command in your extensions.conf use Lame or Sox to do the conversion for you.Cheers,MICHAEL TOOPTel  011 602 9309Fax  011 656 1342
Mobile  083 364 2370Web  www.bizcall.co.zaKuniyoshi Murata wrote: Hi * users, Is that possible to make voicemail audio file (that is attached to
 forwarding email) as MP3 file, rather than WAV? TIA Kuni -- Kuniyoshi Murata English-Japanese Interpreter  Macintosh Webcast Specialist
 [WebSite] www.macwebcaster.com [Email] [EMAIL PROTECTED] [Skype] kuniyoshi_murata [SNS] 
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Re: [Asterisk-Users] A2billing with Mysql-5.0.15

2005-11-14 Thread Administrator TOOTAI


Rafael R. GV a écrit :


[...]

2.-  Change value FG_TABLE_NAME=call t1  for FG_TABLE_NAME=calls 
t1 in following files:


A2Billing_UI/Public/call-log-customers.php
A2Billing_UI/Public/invoices.php
A2BCustomer_UI/balance.php
A2BCustomer_UI/invoices.php


You forgot:

A2Billing_UI/Public/asterisk-stat-v2/
   graph_stat.php
   graph_hourdetail.php
   graph_statbar.php
   graph_pie.php
   call-comp.php
   call-daily-load.php
   call-last-month.php
--
Daniel
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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-14 Thread Sean Cook

Easy: 
 show g729

This will show total in use and total available channels for g729



 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel
 Sent: Monday, November 14, 2005 6:55 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] How to check how many G729 codec
 licenseinstalled
 
 On Mon, 2005-11-14 at 12:22 +0100, Gentian Bajraktari wrote:
  Dependent on what Channel you are using..
  If you are using SIP then do:
 
  *CLI sip show channels
 
 
 That can get really ugly if you have multiple channels in use.  Asterisk
 supports (afaik) 5 out of the box and 6 total VoIP protocols.  Which
 means that you would have to perform and parse for each supported
 protocol.
 
 The manager interface may have what you want, I havent looked at it
 enough to know, and this type of question almost makes me think that its
 for a program to auto parse the information than it is for human
 consumption.  Given that the manager interface is more likely a better
 target to get the information from.
 
 
 --
 Trixter http://www.0xdecafbad.com Bret McDanel
 UK +44 870 340 4605   Germany +49 801 777 555 3402
 US +1 360 207 0479 or +1 516 687 5200
 FreeWorldDialup: 635378

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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
 Easy: 
  show g729
 
 This will show total in use and total available channels for g729

doesnt work for me, maybe its a version difference.

I do have g729 loaded, and that was verified.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Linksys PAP2: supported codecs

2005-11-14 Thread Paul Hayes





yes that's what i'm lead to believe as well. Only the SPA-2100 
SPA-2002 support two simultaneous g.729 calls, the older/lesser models
don't have the processing power required to encode two g.729 streams.

Rich Adamson wrote:

  
I don't think they want to solve it. It's the same with the Sipura boxes.
Only SPA 2100 supports 2 G729 sessions.


  
  
The archives suggest the original models didn't have enough processing
power to handle the compute-intensive g729 codec. I'd have to guess
that is a correct assessment from what I've seen.


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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-14 Thread Andreas Sikkema

 On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
  Easy: 
   show g729
  
  This will show total in use and total available channels for g729
 
 doesnt work for me, maybe its a version difference.
 
 I do have g729 loaded, and that was verified.

Do you have the Digium G.729 codec installed? This one provides show
g729


I have no idea if the IPP hack provides a similar interface.


-- 
Andreas Sikkema   BBned NV
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp 
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Re: [Asterisk-Users] Asterisk Installation exits with following error ***

2005-11-14 Thread BJ Weschke

 You need the libidn-devel package installed.

On 11/14/05, Zeeshan [EMAIL PROTECTED] wrote:
 How do I install curl?

 Zeeshan A Zakaria


 -Original Message-
 From: Rich Adamson [mailto:[EMAIL PROTECTED]
 Sent: Sunday, November 13, 2005 10:41 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk Installation exits with following
 error


  After running make clean; make install, asterisk starts installing
 itself but then
 terminates after some time with the following error:
 
 
 
  /usr/bin/ld: cannot find -lidn
 
  collect 2: Id returned 1 exit status
 
  make[1]: *** [app_curl.so] Error 1
 
  make[1]: Leaving directory '/usr/src/asterisk/apps'
 
  make: *** [subdirs] Error 1
 
 
 
 
 
  On typing asterisk at command prompt doesn't start asterisk, which
 mean it didn't
 install successfully. I never had this problem
  previousely, what am I mississing this time. Did I miss something when
 installing
 Fedora Core 3?
 

 The message is suggesting you don't have 'curl' installed. (I had the
 same issue some time ago with fc3.)

 Download curl (and if I'm not mistaken, you need the curl source code
 installed as well).

 Then do another 'make clean' followed by 'make install'.


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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

On Mon, 2005-11-14 at 13:57 +0100, Andreas Sikkema wrote:
  On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
   Easy: 
show g729
   
   This will show total in use and total available channels for g729
  
  doesnt work for me, maybe its a version difference.
  
  I do have g729 loaded, and that was verified.
 
 Do you have the Digium G.729 codec installed? This one provides show
 g729
 
 
 I have no idea if the IPP hack provides a similar interface.
 
 

Actually I dont have either I have a 3rd option, so it is a variant
difference rather than a version difference.


-- 
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UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Can't create iax channel

2005-11-14 Thread Dinesh Nair




On 11/10/05 15:02 Wayne Gemmell said the following:

When trying to call from this side to that side I get the following

-- Executing Dial(SIP/301-2d50, 
IAX2/wayne:[EMAIL PROTECTED]/204) in new stack
Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any 
of 0xf800 formats
Nov 10 08:37:21 WARNING[30785]: channel.c:455 ast_best_codec: Don't know any 
of 0xf800 formats
Nov 10 08:37:21 WARNING[30785]: chan_iax2.c:7745 iax2_request: Unable to 
create translator path for unknown to ulaw on IAX2/wayne-5


there's your problem right there. what codecs are the SIP peer set to use ? 
apparently, asterisk cant translate between ulaw and the unknown codec.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
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| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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Re: [Asterisk-Users] Can't create iax channel

2005-11-14 Thread Dinesh Nair




On 11/10/05 17:36 Wayne Gemmell said the following:

On Thursday 10 November 2005 10:55, Jason Walker wrote:


The statement of zaptel being required is strange...I use IX trunking
exclusively for my servers. Two of them have no zaptel/Digium hardware and
the trunk calls are fine.


I don't know where I read it, apparently it is needed for timing or something, 
could be in the old handbook or hitchikers guide to asterisk as I havn't got 
far enough into the new handbook to comment.


IAX trunking works even without digium cards as long as the ztdummy pseudo 
timer module is loaded.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo The opinions here in no way reflect the opinions of my $a $b.  |
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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

show g729


  From: Mark Quitoriano [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] How to check how many G729 codec 
licenseinstalled
  Date: Mon, 14 Nov 2005 19:13:29 +0800 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com


 how can i check how many g729 are being used right now?
 
 On 11/13/05, Gentian Bajraktari [EMAIL PROTECTED]  wrote:
 
 Yes.
 
 - Original Message -
 From: Angelito Manansala  [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com 
 Sent: Sunday, November 13, 2005 1:23 PM
 Subject: Re: [Asterisk-Users] How to check how many G729 codec
 licenseinstalled
 
  g723   gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex  ilbc
g723 - - - - - - - - - - -
 gsm - - 3 3 4 3 2 9 - -   131
ulaw - 5 - 1 3 2 1 8 - -   130
alaw - 5 1 - 3 2 1 8 - -   130
g726 - 6 3 3 - 3 2 9 - -   131
   adpcm - 5 2 2 3 - 1 8 - -   130
slin - 4 1 1 2 1 - 7 - -   129
   lpc10 - 8 5 5 6 5 4 - - -   133
g729 - - - - - - - - - - -
   speex - - - - - - - - - - -
ilbc - 9 6 6 7 6 512 - - -
 
 
  this means i have no g729 codec installed..
 
  thanks guys!
 
  :p
 
 
  On 11/13/05, Gentian Bajraktari  [EMAIL PROTECTED] wrote:
  Do:
  *CLI show translations
 
  If you see - (lines) on the G729 row/columns than you do not have any
  G729
  support.
 
 
  RG.
 
  - Original Message -
  From: Sahil Gupta [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Sunday, November 13, 2005 1:03 PM
  Subject: Re: [Asterisk-Users] How to check how many G729 codec
  licenseinstalled
 
 
   Right :)
  
   Regards,
  
  
   Sahil Gupta
   VoiceValley
  
   On Sun, 13 Nov 2005, Angelito Manansala wrote:
  
   *CLI show g729
   No such command 'show g729' (type 'help' for help)
  
   this means i have no g729 codec installed, right?
  
   On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:
   That's easy...
   Just go into asterisk cli and type   show g729  
   It will tell you how many are active and how many you have in total
  
  
   Regards
   Zafer
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto: [EMAIL PROTECTED] On Behalf Of
   Angelito
   Manansala
   Sent: Sunday, 13 November 2005 10:31 PM
   To: asterisk-users@lists.digium.com
   Subject: [Asterisk-Users] How to check how many G729 codec license
   installed
  
   Guys, is the any CLI commands or info files where you can check how
   many g729 codec
   license installed.
  
  
   Regards,
   Lito
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[Asterisk-Users] IAXy echo?

2005-11-14 Thread Mike Hammett




I've got two customers on the same broadband 
provider. Same Asterisk box on my end. Same CLEC.

One has an IAXy and the other has an Asterisk box 
with an array of devices (Grandstream, Cisco, ATCOM, xten, etc.).

The people behind the Asterisk box have had no 
audio quality issues. The person with the IAXy often encounters an 
echo. The echo is only heard on the remote side and it only contains the 
remote caller's voice. This echo has been heard with the remote side being 
varying LECs. The echo is not always there. I'd almost say that the 
echo is not there more than it is.

Troubleshooting next step?

I haven't changed out the IAXy because I don't have 
any other ATAs to put in place.


Mike HammettIntelligent Computing 
Solutionshttp://www.ics-il.com


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RE: [Asterisk-Users] Snom clients deregistering

2005-11-14 Thread Michael Crown

Does the phone ocasionally prompt the user for a password? -Mike

 -Original Message-
 From: Richard Watson [mailto:[EMAIL PROTECTED] 
 Sent: Monday, November 14, 2005 5:00 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Snom clients deregistering

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Hi all,

 I have a server currently running Asterisk 1.0.7 placed out 
 in the wild (i.e. not behind NAT).

 I have groups of sip clients all behind various NAT firewalls 
 (mainly adsl routers).

 Up to now I've mainly used Sipuras and not had any serious problems.
 Recently I've been experimenting with Snom phones and I have 
 encountered  problems where the Snoms register fine initially 
 but after a while (which could be anything from 2minutes to 
 45 minutes) they lose their registration. Sample snom 
 configuration in sip.conf follows:

 [888120]
 type=friend
 username=888120
 mailbox=888120
 canreinvite=no
 nat=yes
 secret=secret
 host=dynamic
 qualify=yes
 context=sipdemo
 subscribecontext=sipdemo

 I've experimented with several different adsl routers and was 
 surprised at the difference this can make, however the 
 problem is still there to a greater or lesser extent.

 I've also tried using a Stun server following recommendation here:

 http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_aud
 io_asterisk.html

 Again this makes a difference, but doesn't entirely solve the 
 problem - there are still occasions where the Snom is 
 unreachable or unknown.

 The implication seems to be that if asterisk does not send 
 keepalives often enough then the way through the nat is lost.

 I've also tried lowering the expiry time of the asterisk 
 sessions (in increments down to 30 seconds) in the hope that 
 it would result in more activity and keep the firewall open, 
 but it didn't help.

 Another strange factor is using the BLF on snoms - the 
 situation seems to be worse with those enabled, but that 
 might not be relevant.

 So I guess I have a few questions:

 1) Has anyone had this happen before and what, if any, was 
 the solution?

 2) How do I increase the frequency with which asterisk sends 
 keepalives?

 3) Does SER handle this better - would placing this outside 
 the NAT help handle connections from inside?

 4) Do newer versions of asterisk handle this better?

 5) Any other suggestions?

 TIA.

 - --
 Richard Watson
 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.1 (GNU/Linux)
 Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

 iD8DBQFDeGAzP05lUVhVYk0RAkM1AKCepBdfTkLoqwNlnbMpH3CWGTWCcwCeOFlE
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Re: [Asterisk-Users] Snom clients deregistering

Michael Crown wrote:
 Does the phone ocasionally prompt the user for a password? -Mike

Yes it does

How did you know?

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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

 Easy: 
  show g729
 
 This will show total in use and total available channels for g729
 doesnt work for me, maybe its a version difference.
 I do have g729 loaded, and that was verified.

Using cvs-head...

If you have the digium licensed g729, the 'show g729' looks like:
 show g729
 0/0 encoders/decoders of 6 licensed channels are currently in use

If you loaded a different g729 codec (unlicensed, but available on the
internet), the response will be No such command...


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Re: [Asterisk-Users] IAXy echo?

2005-11-14 Thread Sergey Okhapkin

Lower speaker volume on the phone connected to IAXy.

On Mon, 2005-11-14 at 07:21 -0600, Mike Hammett wrote:
 I've got two customers on the same broadband provider.  Same Asterisk
 box on my end.  Same CLEC.
  
 One has an IAXy and the other has an Asterisk box with an array of
 devices (Grandstream, Cisco, ATCOM, xten, etc.).
  
 The people behind the Asterisk box have had no audio quality issues.
 The person with the IAXy often encounters an echo.  The echo is only
 heard on the remote side and it only contains the remote caller's
 voice.  This echo has been heard with the remote side being varying
 LECs.  The echo is not always there.  I'd almost say that the echo is
 not there more than it is.
  
 Troubleshooting next step?
  
 I haven't changed out the IAXy because I don't have any other ATAs to
 put in place.
  
  
 
 Mike Hammett
 Intelligent Computing Solutions
 http://www.ics-il.com
  
  
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RE: [Asterisk-Users] Snom clients deregistering

2005-11-14 Thread The VoIP Connection

There is a setting on the Advanced page called Challenge Response on
Phone. Turn this setting to Off and your problem will be solved. Also, we
usually set the Proposed Expiry to 1 minute On the SIP page when phones
are behind a NAT.

-Mike 

 -Original Message-
 From: Richard Watson [mailto:[EMAIL PROTECTED] 
 Sent: Monday, November 14, 2005 8:30 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
 Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Snom clients deregistering
 
 Michael Crown wrote:
  Does the phone ocasionally prompt the user for a password? -Mike
 
 Yes it does
 
 How did you know?
 

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Re: [Asterisk-Users] IAXy echo?


 I've got two customers on the same broadband provider.  Same Asterisk box on 
 my end.  
Same CLEC.
  
 One has an IAXy and the other has an Asterisk box with an array of devices 
(Grandstream, Cisco, ATCOM, xten, etc.).
  
 The people behind the Asterisk box have had no audio quality issues.  The 
 person with 
the IAXy often encounters an echo. 
 The echo is only heard on the remote side and it only contains the remote 
 caller's 
voice.  This echo has been heard with the
 remote side being varying LECs.  The echo is not always there.  I'd almost 
 say that 
the echo is not there more than it is.
  
 Troubleshooting next step?
  
 I haven't changed out the IAXy because I don't have any other ATAs to put in 
 place.

Best guess... the iaxy doesn't have an echo can in it, and probably relies
on asterisk to do the cancellation.


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Re: [Asterisk-Users] IAXy echo?

2005-11-14 Thread [EMAIL PROTECTED]





You will also experience this if the latency between the Asterix PABX
and IAXy is so high that echo cancel don't work.

Jan
Rich Adamson wrote:

  
I've got two customers on the same broadband provider.  Same Asterisk box on my end.  

  
  Same CLEC.
  
  
 
One has an IAXy and the other has an Asterisk box with an array of devices 

  
  (Grandstream, Cisco, ATCOM, xten, etc.).
  
  
 
The people behind the Asterisk box have had no audio quality issues.  The person with 

  
  the IAXy often encounters an echo. 
  
  
The echo is only heard on the remote side and it only contains the remote caller's 

  
  voice.  This echo has been heard with the
  
  
remote side being varying LECs.  The echo is not always there.  I'd almost say that 

  
  the echo is not there more than it is.
  
  
 
Troubleshooting next step?
 
I haven't changed out the IAXy because I don't have any other ATAs to put in place.

  
  
Best guess... the iaxy doesn't have an echo can in it, and probably relies
on asterisk to do the cancellation.


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RE: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-14 Thread Sean Cook

Are you running the g729 module from digium?  Registered?

Sean

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel
 Sent: Monday, November 14, 2005 7:50 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] How to check how many G729 codec
 licenseinstalled
 
 On Mon, 2005-11-14 at 07:46 -0500, Sean Cook wrote:
  Easy:
   show g729
 
  This will show total in use and total available channels for g729
 
 doesnt work for me, maybe its a version difference.
 
 I do have g729 loaded, and that was verified.
 
 --
 Trixter http://www.0xdecafbad.com Bret McDanel
 UK +44 870 340 4605   Germany +49 801 777 555 3402
 US +1 360 207 0479 or +1 516 687 5200
 FreeWorldDialup: 635378

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[Asterisk-Users] Configure Asterisk to call from softPhone(SIP Channel) to Analog phone(Modem Channel)

2005-11-14 Thread ashok


Hi *users,,

I'm researching on Asterisk PBX phone system initially I was successfull in
configuring 2 SIP users with DIAL rules in extension.conf and
configured 2X-Lite softphones to use my proxy
Registered successfully also able to dial and communicate.

Now i am trying to dial from softphone to analog phone
connected to Internal Modem of my proxy but ended up
with errors while loading asterisk -gc

Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so] = (Generic Voice Modem Driver)
  == Parsing '/etc/asterisk/modem.conf': Found
  == Loading modem driver chan_modem_slamr.soNov 14
15:02:15 WARNING[8042]: loader.c:258
ast_load_resource:
/usr/lib/asterisk/modules/chan_modem_slamr.so: cannot
open shared object file: No such file or directory
Nov 14 15:02:15 ERROR[8042]: chan_modem.c:968
load_module: Failed to load driver chan_modem_slamr.so
  == Unregistered channel type 'Modem'
Nov 14 15:02:15 WARNING[8042]: loader.c:345
ast_load_resource: chan_modem.so: load_module failed,
returning -1
  == Unregistered channel type 'Modem'
Nov 14 15:02:15 WARNING[8042]: loader.c:391
load_modules: Loading module chan_modem.so failed!

Any idea how to generate chan_modem_slamr.so file???

[EMAIL PROTECTED] slmodem-2.9.10]# more
/etc/modules.conf
alias eth0 8139too
alias eth1 via-rhine
alias usb-controller ehci-hcd
alias usb-controller1 usb-uhci
alias sound-slot-0 via82cxxx_audio
post-install sound-slot-0 /bin/aumix-minimal -f
/etc/.aumixrc -L /dev/null 21 || :
pre-remove sound-slot-0 /bin/aumix-minimal -f
/etc/.aumixrc -S /dev/null 21 || :
alias char-major-212 slamr
alias char-major-213 slusb

Internal modem :- Smartlink chipset v.92 internal pci
Modem

Pls suggest me how do I write DIAL rule so that user 2000 registerd to proxy
via Softphone can dial 2001 to analog phone.

Thanks in advance.

Warm Regards
ashok

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RE: [Asterisk-Users] Sipura SPA-2002 Double Ring

  I recently implemented a Sipura SPA-2002 with one of my Asterisk
  installations.  On internal calls, the SPA generates ringtone as
  expected. However, when I dial out via my IAX-based service
  provider, I hear 
  both the telco-generated ringtone as well as the SPA-generated
  ringtone.  Sometimes, the SPA continues to generate the ringtone even
  after the call has been answered.
  
  I don't have a spa-2002, but do use a spa3k. I doubt very much the
  sipura device is actually providing ringback tone, and I don't recall
  any parameters that would enable/disable such an item. (The Admin
  manual does not mention it either.)   
  
  You might check your extensions.conf entry for dialing your provider
  to see if you have an r in that line. If so, remove it. 
 
 The SPA-2002 is definitely generating the additional ringback.  I verified
 this by temporarily changing the frequency of the ringback in the SPA's
 Regional settings.  
 
 I also verified that I am not using the r option in the Dial command.  If
 I were, however, only the Asterisk-generated ringback would be heard, and
 then only until the call supervised (i.e. I would not be hearing two
 distinct ring signals, and the ringback would not occasionally persist for
 the duration of a call while still hearing the called party).  
 
 This problem is present only with the SPA-2002, and none of the other SIP
 devices connected to this Asterisk server.  I have also tried making
 outbound calls via different service providers, all with the same results.  

If I had this problem, I'd use ethereal to observe the sip traffic to
the box and look for a control packet containing RING. If that is
coming from your asterisk box after a call in progress, then asterisk
isn't functioning properly. 

If you don't see that packet, then I'd be on the horn to sipura support.
(Make sure you're running the latest firmware for the box as that will
always be their first suggestion.)



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[Asterisk-Users] OT: Aastra PT 390 Question.

2005-11-14 Thread Richard Reina

Does anyone know how to put an Aastra PT 390 in
headset mode, so it will only give a dial tone when
you are ready ?  Right now I can't figure how to keep
it hung up?  If I hit googbye it merely flashes (give
me a dial tone again).

Any help would be greatly appreciated?




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[Asterisk-Users] SIP signaling and canreinvite=yes

2005-11-14 Thread Damon Estep









After reviewing many other posts as well as wiki information
on canreinvite and asterisk media path I am not clear on whether asterisk still
manages sip signaling after a reinvite has been issued between a peer and a UA.



Here are the details;



UA g.711u Asterisk g.711u SIP long distance
provider.

The SIP LD provider uses a session border controller to
ensure that all sip traffic originates from my asterisk IP address.

The SIP LD provider will accept RTP streams from any source.



Due to an issue when sending faxes with * in the media
stream, I want to remove asterisk from the media stream for specific UAs (faxes
complete successfully without asterisk in the stream, tested by setting the UA
to the asterisk IP address).



In theory, if canreinvite=yes, codecs match (g.711u) and
there are no dial options that require asterisk to remain in the stream, the
re-invite should be issued and the UA and the peer should be the endpoints of
the RTP streams.



Questions;



Does it work? I am having trouble getting it to work that
way.

Is the sip signaling all handled by asterisk in this case? 
required by my providers session border controller.



I guess what I am asking is can asterisk function as a SIP
PROXY when configured correctly?



Any examples or limitations I might have missed?



Thank you!



Damon










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Re: [Asterisk-Users] ISDN card required

2005-11-14 Thread Kristof Hardy


Lee Archer wrote:
Can anyone point me in the direction of a quality, works with Asterisk, 
BRI card.  I need minimum 2 port/4 channel.


Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.

Cheers.

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RE: [Asterisk-Users] ISDN card required

2005-11-14 Thread Lee Archer

Thanks to all.  I'll probably go with the quadBri card they do.

Regards

Lee 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kristof
Hardy
Sent: 14 November 2005 14:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] ISDN card required

Lee Archer wrote:
 Can anyone point me in the direction of a quality, works with 
 Asterisk, BRI card.  I need minimum 2 port/4 channel.

Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.

Cheers.

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Re: [Asterisk-Users] Snom clients deregistering

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

The VoIP Connection wrote:
 There is a setting on the Advanced page called Challenge Response on
 Phone. Turn this setting to Off and your problem will be solved. Also, we
 usually set the Proposed Expiry to 1 minute On the SIP page when phones
 are behind a NAT.

That doesn't seem to have helped entirely.

The Password prompt no longer appears but the phone still becomes
UNREACHABLE then UNKNOWN after a few minutes.

In the system information on the phone it reports Registration Failed.
However a few minutes later it logs itself back in.

I have two identical snoms on the bench here and they both do the same
thing, logging in and operating fine, before eventually (but not
necessarily at the same time) losing registration and stopping for a few
minutes.




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

iD4DBQFDeKPnP05lUVhVYk0RAqTfAJYtZqmp1dCRLDhu3C1jHRCeUk5LAJ42z2rV
5Jr8qm+Ruyvv3h2L3jOjUA==
=PlHs
-END PGP SIGNATURE-
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[Asterisk-Users] Re: MYSQL issue in UPDATE..

2005-11-14 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Mauro Zanin [EMAIL PROTECTED] wrote:
 Hi Everybody,
 I'm trying to execute a MYSQL(UPDATE..) sql
 command over a table I have previously red. I get a timeout and no update
 happens.
 I use  * 1.0.9.
 I wonder if MYSQL set of commands allows Update...

Yes, I use UPDATE within MYSQL() successfully.

If you post the complete extract from your dialplan, starting with the
MYSQL(Connect... up to the MYSQL(Disconnect..., then we might be able
to suggest where the problem lies.

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Brooktrout MPAC 1200 card with Asterisk

2005-11-14 Thread Stephen Arulraj






I have a 4 port brooktrout PCI E1/T1 blade card (MPAC 1200) that
was used for some carrier server. Will Asterisk support this? Has
anyone used this
successfully before? Thanks! Stephen


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[Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk

2005-11-14 Thread nr k

Hi All

Can anybody tell me the maximum number of SIP Phones
supported by Asterisk.

regards
ramakrishnan.n




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[Asterisk-Users] asterisk sample size adjustment

Is there any way to adjust the sample size asterisk uses for VoIP
codecs?  From what I have gathered it uses a fixed 20ms sample size for
all codecs.  While some require at least this, some can be configured
for less.  This results in more overhead, but can be tweaked to provide
more efficient transfer on the backbone links due to ATM framing
properties.

If anyone has any information on how to change the sample size I would
appreciate hearing about it, because I cant find anything with google.
Asterisk is a particularly bad google term since it is used as a
footnote market, wildcard, etc :P


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Can't make calls from Asterisk IAX to other IAX

2005-11-14 Thread chawki hammoud

Sorry, I just saw the post.

Yes, it's the same format

Regards;
Chawki

--- Matt Riddell [EMAIL PROTECTED] wrote:

 chawki hammoud wrote:
  Hi:
  
  I have been having this problem for sometime that
 I am
  not able to solve and I hope someone can help. 
  
  I can make VOIP calls between my Asterisk box and
 my
  VOIP provider using sip channel without a problem.
 But
  when I attempt to make a call using IAX, the call
 get
  accepted and then get a hangup message:
 
 is this the same number format you send when using
 sip: 0017046872001
 
 -- 
 Cheers,
 
 Matt Riddell
 ___
 
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 News - html)
 http://freevoip.gedameurope.com (Free Asterisk Voip
 Community)
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 News - rss)
 
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[Asterisk-Users] connect to gateway h323

2005-11-14 Thread Reli Loin

Hello,

Hello, I do not arrive has to connect me has a gateway h323, in
termination of call.

i have one ip for a termination call xxx.xx.xx.xx,

I do not know if the problem comes from my parameters oh323.conf or the gateway


i using a latest version asterisk (asterisk 1.2rc1),openh323 latest
version mimas patch,
and pwlib latest version and asterisk-0h323-0.7.3

my config files.

-oh323.conf---

h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=100
inboundMax=100
simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.

; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=0
libTraceLevel=0
libTraceFile=stdout
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is
the zone name.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   gatekeeper's DNS name,
;   gatekeeper's ip,
;   GKID:gatekeeper's id
;   gatekeeper's id@gatekeeper's name or address
;
;gatekeeper=192.168.1.6
gatekeeper=xxx.xxx.xxx.xx
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=60
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;   INBAND  -
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=h323

;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=asterisk
alias=123
;
; Aliases/prefixes routed in all-aliases context.
;
context=all-aliases
alias=ASTERISK
alias=666
;
; Aliases/prefixes routed in more-aliases context.
;
context=more-aliases
alias=665
;
; Aliases/prefixes routed in all-prefixes context.
;
context=all-prefixes
gwprefix=00
gwprefix=01
;


; Aliases/prefixes routed in more-stuff context.
;
context=more-stuff
alias=664
gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every codec option may have a frames option
; associated with it.
; Valid values for the codec option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G726-   G.726(32k)
;   G72616K -   G.726(16k)
;   G72624K -   G.726(24k)
;   G72632K -   G.726(32k)
;   G72640K -   G.726(40k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10   -   LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
codec=G711A
frames=20
codec=G711U
frames=20
;codec=GSM0610

;frames=4
;codec=G7231
;frames=2
codec=G729
frames=2

[h323terminate]
type=peer
host=xx.xxx.xxx.xxx
dtmfcodec=99

--


oh323 show conf in asterisk cli

 Configuration of OpenH323 channel driver
--
Version: 0.7.3
Listening on address: 0.0.0.0:1720
Gatekeeper used:  No gatekeeper
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref.

Re: [Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk

On Mon, 2005-11-14 at 06:53 -0800, nr k wrote:
 Hi All
 
 Can anybody tell me the maximum number of SIP Phones
 supported by Asterisk.

If I run asterisk on my ipaq not very many.  If I run it on a real
server many many more.

Your question cant really be answered with the information you have
provided.  It can only be answered in context.

What hardware?

What codecs?

Any translations (from one medium/codec to another)?

What applications are used (AGI, conferences, voicemail, etc)?

Is the asterisk server actually pushing the bits for a call or just
doing call setup and connecting the two endpoints directly?


These are the very minimum questions you have to answer before your
question can be answered.  There are a few other things that can go into
it, but those will help you better define for a rough idea ...  And
based on the answers to those questions there may be more questions.

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Asterisk behind a NAT

2005-11-14 Thread Martinez Felix

You need a Stun Server...vovida.org works for meOn 11/11/05, Enrique Leon [EMAIL PROTECTED]
 wrote:Second postI have installed Asterisk on SuSE 10.0 with an active firewall/NAT filter.
The server has connection to my own Intranet (private IP) and to InternetEverything works well for clients behind and in front-of the firewallbut they can not communicate with each other. Signalling gets through
but the audio gets blocked by the firewall/NAT.So, I open-up ports 10.000 -to- 20.000 in the fw so that the udp/rtp packagescuold get through but it has not been successful.I am using xlite for clients and have no pot cards installed ( digium
fxo,fxs, etc).Does anyone knows what else to do?Has anyone come accross (and solved) this type of problem?Firewall configuration is as follows:FW_DEV_EXT=eth-id-00:0d:87:5c:44:e5 #eth1
FW_DEV_INT=eth-id-00:06:4f:0e:ca:99eth-id-00:40:f4:9f:12:25 #eth0 wlan0FW_ROUTE=yesFW_MASQUERADE=yesFW_MASQ_DEV=$FW_DEV_EXTFW_MASQ_NETS=
192.168.100.0/255.255.255.0FW_SERVICES_EXT_TCP=53 http https sshFW_SERVICES_EXT_UDP=5060 5061 53FW_SERVICES_INT_TCP=21 3128 5056 53 5801 5901 80 8080epmap http microsoft-ds netbios-ssn smtp ssh
FW_SERVICES_INT_UDP=5060:5075 53 bootps netbios-dgmnetbios-nsFW_SERVICES_INT_RPC=mountd nfs nfs_acl nlockmgrportmap status ypbindFW_SERVICES_ACCEPT_EXT=0/0,udp,5060:5075
FW_TRUSTED_NETS=192.168.100.0/255.255.255.0FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,5060
FW_FORWARD=0/0,192.168.100.0/255.255.255.0,udp,1FW_FORWARD=192.168.100.0/255.255.255.0,0/0,udp,1
Sip Configuration:[general]bindport=5060bindaddr=0.0.0.0srvlookup=noexternrefresh=10externip=201.208.246.178
nat=yeslocalnet=192.168.100.0/255.255.255.0;RTP configuration:[general]rtpstart=1rtpend=2rtpchecksums=yesRegards, Enrique Leon
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[Asterisk-Users] Problem with 827-4v and asterisk as a pstn GW

2005-11-14 Thread Simone Ricci

Hi,
I've a problem with a cisco 827-4v and asterisk (1.0.9) acting as
sip-to-pstn GW. The issue is that when a call comes in from the pstn,
asterisk correctly contacts the router, which in turns send a 183
Session progress. Obviously, asterisk thinks that the telephone is not
ringing (because it expects a 180 Ringing) and we have no ringback on
the pstn side. Putting a ringing() in the dialplan is not an option.

Anyone has suggestions?

Cheers,
Simone.
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Re: [Asterisk-Users] ISDN card required

2005-11-14 Thread Klaus Darilion


Kristof Hardy wrote:

Lee Archer wrote:

Can anyone point me in the direction of a quality, works with 
Asterisk, BRI card.  I need minimum 2 port/4 channel.



Ack. Like Mark pointed out, I also used Junghanns.net cards, works fine.


Hi Kristof!

(sorry for the empty email)

Do you use it with asterisk 1.2 (CVS)? AFAIK the bristuff package for 
1.2 is quiet out-of-date.


btw: have you ever used chan_misdn from beronet with quadBRI cards? Any 
experiences?


regards
klaus
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[Asterisk-Users] problem to connect h323 temination

2005-11-14 Thread Reli Loin

Hello,

Hello, I do not arrive has to connect me has a gateway h323, in
termination of call.

i have one ip for a termination call xxx.xx.xx.xx,

I do not know if the problem comes from my parameters oh323.conf or the gateway


i using a latest version asterisk (asterisk 1.2rc1),openh323 latest
version mimas patch,
and pwlib latest version and asterisk-0h323-0.7.3

my config files.

-oh323.conf---

h245Tunnelling=yes
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=yes
;
; Set jitter buffer (in milliseconds, 20...1).
;
jitterMin=20
jitterMax=100
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
;   lowdelay, throughput, reliability, mincost, none
; Moreover, an integer (in decimal or hex format) may be entered.
;
ipTos=none
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=100
inboundMax=100
simultaneousMax=100
;
; Call Rate Limiter params (ingress direction). When the total number
; of active calls is above 'crlThreshold' then the rate of the incoming
; H.323 calls is restricted in a way where no more than 'crlCallNumber'
; calls are allowed in 'crlCallTime' milliseconds, thus limiting the rate
; of incoming calls to:
; 'crlCallNumber' / ('crlCallTime' / 1000) Calls-per-Sec.
;
;crlCallNumber=20
;crlCallTime=2
;crlThreshold=30
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.

; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=0
libTraceLevel=0
libTraceFile=stdout
;
; Disable gatekeeper or specify a gatekeeper. The gatekeeper's ID is
the zone name.
; Valid values for this option are:
;   DISABLE,
;   DISCOVER,
;   gatekeeper's DNS name,
;   gatekeeper's ip,
;   GKID:gatekeeper's id
;   gatekeeper's id@gatekeeper's name or address
;
;gatekeeper=192.168.1.6
gatekeeper=xxx.xxx.xxx.xx
;
; Set the gatekeeper password. If used, it enables H.235 access to gatekeeper.
;
;gatekeeperPassword=secret
;
; Set the gatekeeper registration timeout. Before the expiration of
; the timeout, a re-registration is attempted.
;
gatekeeperTTL=60
;
; Set the mode for sending user-input (DTMF)
; Valid values for this option are:
;   Q931-   Q.931 Keypad Information Element
;   STRING  -   H.245 string
;   TONE-   H.245 tone
;   RFC2833 -   RFC2833
;   INBAND  -
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
;
accountCode=H323
;
; Default language
;
language=en
;
; Default Music-On-Hold class
;
musiconhold=default
;
; Set the default context of H.323 calls.
;
context=h323

;-
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-
[register]
;
; Aliases/prefixes associated with the default context
; defined in section [general].
;
alias=asterisk
alias=123
;
; Aliases/prefixes routed in all-aliases context.
;
context=all-aliases
alias=ASTERISK
alias=666
;
; Aliases/prefixes routed in more-aliases context.
;
context=more-aliases
alias=665
;
; Aliases/prefixes routed in all-prefixes context.
;
context=all-prefixes
gwprefix=00
gwprefix=01
;


; Aliases/prefixes routed in more-stuff context.
;
context=more-stuff
alias=664
gwprefix=02

;-
; Specify and configure CODEC related
; options
;-
[codecs]
;
; Define the codec list of the channel driver.
; Every codec option may have a frames option
; associated with it.
; Valid values for the codec option are:
;   G711U   -   G.711 u-Law
;   G711A   -   G.711 A-Law
;   G7231   -   G.723.1(6.3k)
;   G72316K3-   G.723.1(6.3k)
;   G72315K3-   G.723.1(5.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G7231A6K3   -   G.723.1A(6.3k)
;   G726-   G.726(32k)
;   G72616K -   G.726(16k)
;   G72624K -   G.726(24k)
;   G72632K -   G.726(32k)
;   G72640K -   G.726(40k)
;   G728-   G.728
;   G729-   G.729
;   G729A   -   G.729A
;   G729B   -   G.729B
;   G729AB  -   G.729AB
;   GSM0610 -   GSM 0610
;   MSGSM   -   Microsoft GSM Audio Capability
;   LPC10   -   LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
codec=G711A
frames=20
codec=G711U
frames=20
;codec=GSM0610

;frames=4
;codec=G7231
;frames=2
codec=G729
frames=2

[h323terminate]
type=peer
host=xx.xxx.xxx.xxx
dtmfcodec=99

--


oh323 show conf in asterisk cli

 Configuration of OpenH323 channel driver
--
Version: 0.7.3
Listening on address: 0.0.0.0:1720
Gatekeeper used:  No gatekeeper
FastStart/H245Tunnelling/H245inSetup: ON/ON/ON
Supported formats in pref.

[Asterisk-Users] Comments in AEL files?

2005-11-14 Thread Ed Greenberg

Any way to comment out a line (or some text) in an AEL file? 
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Re: [Asterisk-Users] Snom clients deregistering

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Michael Crown wrote:
 Did you change the proposed expiry? -Mike 

Yes, now set to 1 minute.

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

iD8DBQFDeLLTP05lUVhVYk0RAuHeAJwOio/yEfblrUEnIaQsjXVbaqdj8gCfQfMC
FjPmGjtICurLTdN9DAiXQVg=
=JgQF
-END PGP SIGNATURE-
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Re: [Asterisk-Users] Comments in AEL files?

2005-11-14 Thread Sergey Okhapkin

//comment

AEL ignores any text from // till the line end.

On Mon, 2005-11-14 at 07:39 -0800, Ed Greenberg wrote:
 Any way to comment out a line (or some text) in an AEL file? 
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Re: [Asterisk-Users] asterisk sample size adjustment

2005-11-14 Thread [EMAIL PROTECTED]





hi,

723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
'standard' for 711 is actually 6ms (48 bytes). This would have to be
done per channel (or per codec), but I am not sure wherever Asterisk
allow per codec size or run's with one static size???

Jan

trixter aka Bret McDanel wrote:

  Is there any way to adjust the sample size asterisk uses for VoIP
codecs?  From what I have gathered it uses a fixed 20ms sample size for
all codecs.  While some require at least this, some can be configured
for less.  This results in more overhead, but can be tweaked to provide
more efficient transfer on the backbone links due to ATM framing
properties.

If anyone has any information on how to change the sample size I would
appreciate hearing about it, because I cant find anything with google.
Asterisk is a particularly bad google term since it is used as a
footnote market, wildcard, etc :P


  
  

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Re: [Asterisk-Users] asterisk sample size adjustment

On Mon, 2005-11-14 at 16:57 +0100, [EMAIL PROTECTED] wrote:
 hi,
 
 723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
 'standard' for 711 is actually 6ms (48 bytes). This would have to be
 done per channel (or per codec), but I am not sure wherever Asterisk
 allow per codec size or run's with one static size???

Yeah and the gsm one usually uses 20ms.  A per codec way would be ideal,
I implied that in my original post, or something.  I just thought that
for network tuning purposes it might be nice to actually have that
ability.  Less padding more payload on the ATM cells makes for a more
efficient network :)

AFAIK all the sample sizes are hardcoded, but figured I would ask and
see if anyone knew of a way short of altering the code to adjust this.
While altering the code is usually not a problem, it makes updating a
little more work and stuff..


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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[Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

2005-11-14 Thread Amaury BOSSE









Hi all,



I am trying to install an AVM Fritz card USB v2.1 on
my Asterisk Box.

I am using Debian Sarge with 2.6.8 kernel.

I have compiled capi last drivers (fcusb2-suse93-3.11-07.tar.gz)
and have copied fcusb.ko to /lib/modules/2.6.8/extra/.



All modules seems loaded (capi, capifs, kernelcapi,
fcusb,)

Capiinfo fails and returns : capi
not installed - No such device or address (6)



I have tried to install chan_capi-cm-0.6.1.tar.gz but
Asterisk no longer starts.

/var/log/asterisk/messages returns :

Nov 14 16:40:51 WARNING[4005]: CAPI not installed,
CAPI disabled!

Nov 14 16:40:51 WARNING[4005]: chan_capi.so:
load_module failed, returning -1

Nov 14 16:40:51 WARNING[4005]: Loading module
chan_capi.so failed!



I have tried to find out a solution from the web but
without results.

Does someone know where the problem is from?



Thanks for your help

Amaury






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[Asterisk-Users] TDM400 cards and modem/fax devices

2005-11-14 Thread Chris Bagnall

Hi all,

Having read the various fax and asterisk pages on voip-info, am I right in
thinking I should be able to bridge Zap channels carrying fax without
reliability problems (which as I understand things plague Fax-over-IP)?

The reason for asking is in relation to a requirement where both fax and an
antiquated EPOS system need use of an analogue line - the fax for both
sending and receiving, the EPOS purely for dialling out to shops. The site
has 2 analogue lines for this purpose, each with different numbers, both of
which are listed as fax numbers for the 2 companies in question. Fax volume
is low, so 2 separate fax machines would be uneconomical.

Would it be feasible to feed the 2 analogue lines into a TDM400 on FXO
modules, then connect the EPOS and fax to FXS modules on the same card? I
assume it'd then be possible to route all incoming calls on either line to
the fax machine, and allow the EPOS system to dial out using whichever line
it chooses.

Questions:
1) Would this approach be sufficiently reliable to ensure faxes weren't
lost?
2) Is there a simpler way of accomplishing this without complicating things
with the telephone company?

Regards,

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


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RE: [Asterisk-Users] spandsp-0.0.2pre21c broken?

2005-11-14 Thread Anton Krall

If you patch rc1's makefile manually, you can compile spandsp without
problems, but if you try the same thing with rc2, you'll notice that spandsp
seems to be broken against rc2.

Waiting for Steve to shed some light on this.
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Lee Archer
|Sent: Monday, November 14, 2005 2:27 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: RE: [Asterisk-Users] spandsp-0.0.2pre21c broken?
|
|I get an error when patching the makefile, seems the order is 
|different.
|Had the same problem with rc1 and 2. 
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: 13 November 2005 17:34
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] spandsp-0.0.2pre21c broken?
|
|Guys. Has anybody been able to compile spandsp-0.0.2pre21c 
|against 1.2rc2?
|
|Seems spandsp-0.0.2pre21c is broken. :(
|
|Compiles great against 1.2rc1 but no luck so far with rc2.
|
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|###
|
|This message has been scanned by F-Secure Anti-Virus for 
|Microsoft Exchange.
|For more information, connect to http://www.f-secure.com/ 
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[Asterisk-Users] PRI to SIP

2005-11-14 Thread FaberK

Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
ast*CLI
-- Executing SetCallerID(Zap/14-1, outside) in new stack
-- Executing Set(Zap/14-1, CALLERID=outside) in new stack
-- Executing Dial(Zap/14-1,
SIP/[EMAIL PROTECTED]:5060|30|r) in new stack
-- Accepting call from 'outside' to 'mynumber201' on channel 0/14, span 1
-- Called [EMAIL PROTECTED]:5060
Nov 14 17:15:47 NOTICE[8459]: chan_sip.c:9492 handle_response_invite:
Failed to authenticate on INVITE to 'Unknown
sip:[EMAIL PROTECTED];tag=as6261e060'
-- SIP/sip.mydomain.com:5060-5eda is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion(Zap/14-1, ) in new stack
-- Channel 0/14, span 1 got hangup request
  == Spawn extension (default, 020201, 4) exited non-zero on 'Zap/14-1'
-- Hungup 'Zap/14-1'
ast*CLI
--
my extensions:
--
[general]
static=yes
writeprotect=no

[globals]
;TRUNK=Zap/g2
;TRUNKMSD=1
PRITRUNK1=Zap/g1
PRITRUNK2=Zap/g2
MYNUM=1234567

[default]

exten = _1234567XXX,1,SetCallerID(${CALLERID})
exten = _1234567XXX,2,Set(CALLERID=${CALLERID})
exten = _1234567XXX,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED]:5060,30,r)
exten = _1234567XXX,103,Hangup
--
Where I'm wrong?
What's missing?
Thanks!
--
.:FaberK:.
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RE: [Asterisk-Users] Zaptel cards on SuSE?

2005-11-14 Thread Michael West

Hi Ramon,

I have used Asterisk 1.09, SuSE 9.3 with a TDM400M with 4 FXOs.  I'm
planning on trying 10, but haven't found the time. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, November 13, 2005 9:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Zaptel cards on SuSE?

Hello:

So far I have been using Asterisk with SIP and VoIP only.

I just received a couple a Zaptel cards from Digium (one analog 2 FXS +
2 FXO, one T1), but I am hesitant to install them because I am afraid I
may break the kernel or something.

Since Asterisk is not tested under SuSE, I prefer to proceed with
caution.

So, is there anyone out there using the Zaptel cards under SuSE?

TIA,

-Ramon F Herrera

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Re: [Asterisk-Users] spandsp-0.0.2pre21c broken?

2005-11-14 Thread Eric \ManxPower\ Wieling


Anton Krall wrote:

If you patch rc1's makefile manually, you can compile spandsp without
problems, but if you try the same thing with rc2, you'll notice that spandsp
seems to be broken against rc2.

Waiting for Steve to shed some light on this.


As far as I know spandsp does NOT require Asterisk and does NOT build 
against asterisk.  Only rx_fax and tx_fax do that, and they are not 
part of the spandsp package.

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[Asterisk-Users] How do I know if I have CRC-CCITT (README.Linux26)

2005-11-14 Thread Chuck Bunn


Hi,

I have downloaded the latest release candidate v1-2-0-rc2 and I was 
checking the readme's in the zaptel directory and I came across a 
requirement I have not seen before. In the readme it says that for Linux 
2.6 kernel you will need to have the 'CRC-CITT' functions compiled with 
the kernel. I am using Fedora 4 kernel version 2.6.13-1.1532.FC4. How 
can I verify that I have these functions?


Thanks
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[Asterisk-Users] NAT setup

2005-11-14 Thread Andre Courchesne - Consultant


Hi all,

 I am setting up a a proof on concept where a SIP phone sits on the 
internet and connects to a * behing a NAT.


 Right now the SIP phone connects to the * box just fine, I can dial 
and I see the commands being executed on the * box, but I don't have any 
audio on the SIP phone. Any idas/pointers?


Thanks,

Andre Courchesne
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Re: [Asterisk-Users] How do I know if I have CRC-CCITT (README.Linux26)

2005-11-14 Thread Saul Diaz


Chuck Bunn wrote:

You have

lsmod  | egrep crc_ccitt

check in your kernel modules looking for crc_ccitt

but FC4 comes with that

regards
Saul


Hi,

I have downloaded the latest release candidate v1-2-0-rc2 and I was 
checking the readme's in the zaptel directory and I came across a 
requirement I have not seen before. In the readme it says that for 
Linux 2.6 kernel you will need to have the 'CRC-CITT' functions 
compiled with the kernel. I am using Fedora 4 kernel version 
2.6.13-1.1532.FC4. How can I verify that I have these functions?


Thanks
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Re: [Asterisk-Users] How do I know if I have CRC-CCITT (README.Linux26)

2005-11-14 Thread ram

Hi

i have installed FC4 with select everything
with your command i dont find any thing
results are null
thats means its not installed ??

ram
On 11/14/05, Saul Diaz [EMAIL PROTECTED] wrote:
Chuck Bunn wrote:You havelsmod| egrep crc_ccittcheck in your kernel modules looking for crc_ccitt
but FC4 comes with thatregardsSaul Hi, I have downloaded the latest release candidate v1-2-0-rc2 and I was checking the readme's in the zaptel directory and I came across a
 requirement I have not seen before. In the readme it says that for Linux 2.6 kernel you will need to have the 'CRC-CITT' functions compiled with the kernel. I am using Fedora 4 kernel version 
2.6.13-1.1532.FC4. How can I verify that I have these functions? Thanks ___ --Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Anybody tried it from India ?.



On Nov 14, 2005, at 2:50 AM, Dinesh wrote:

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom  
Rymes

Sent: Monday, November 14, 2005 1:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Anybody tried it from India ?.

On Nov 14, 2005, at 12:37 AM, ram wrote:


Hi

its not legal in india
connecting to PSTN to VOIP

ram


Asterisk doesn't necessarily mean VOIP. He could set it up using ZAP
channels only and not have any VOIP in use at all.

Tom


Its illegal to interconnect it to the local pstn (from abroad).

Dinesh.


I still don't see how this would stop him from using no VOIP  
protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,  
no VOIP, no bypassing the Indian telephone monopoly (assuming that  
there is one).


Tom

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

Technology solutions for small and medium sized businesses.



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RE: [Asterisk-Users] asterisk sample size adjustment

2005-11-14 Thread Dan Austin

 hi,
 
 723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
 'standard' for 711 is actually 6ms (48 bytes). This would have to be
 done per channel (or per codec), but I am not sure wherever Asterisk
 allow per codec size or run's with one static size???

There is a patch in Mantis, bugid 5162, I think, that allows for
changing
the packetization size.  It surrently works on  global/users/peers, but 
is not per codec.

Bug 5162 updates the  core RTP code and adds the packetization options
to chan_sip.  There is a seperate patch to add packetization to the 
ooH323c version of chan_h323.

 Yeah and the gsm one usually uses 20ms.  A per codec way would be
ideal,
 I implied that in my original post, or something.  I just thought that
 for network tuning purposes it might be nice to actually have that
 ability.  Less padding more payload on the ATM cells makes for a more
 efficient network :)

 AFAIK all the sample sizes are hardcoded, but figured I would ask and
 see if anyone knew of a way short of altering the code to adjust this.
 While altering the code is usually not a problem, it makes updating a
 little more work and stuff..

Dan
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RE: [Asterisk-Users] Anybody tried it from India ?.

2005-11-14 Thread Dean Collins

There are a number of asterisk implementations in India.

Cheers,
Dean


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Tom Rymes
 Sent: Monday, November 14, 2005 12:10 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Anybody tried it from India ?.
 
 
 On Nov 14, 2005, at 2:50 AM, Dinesh wrote:
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Tom
  Rymes
  Sent: Monday, November 14, 2005 1:43 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [Asterisk-Users] Anybody tried it from India ?.
 
  On Nov 14, 2005, at 12:37 AM, ram wrote:
 
  Hi
 
  its not legal in india
  connecting to PSTN to VOIP
 
  ram
 
  Asterisk doesn't necessarily mean VOIP. He could set it up using
ZAP
  channels only and not have any VOIP in use at all.
 
  Tom
 
  Its illegal to interconnect it to the local pstn (from abroad).
 
  Dinesh.
 
 I still don't see how this would stop him from using no VOIP
 protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,
 no VOIP, no bypassing the Indian telephone monopoly (assuming that
 there is one).
 
 Tom
 
 --
 Tom Rymes
 Cascade Link Systems
 www.cascadelinksystems.com
 (603) 375-1414
 
 Technology solutions for small and medium sized businesses.
 
 
 
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RE: [Asterisk-Users] Anybody tried it from India ?.

2005-11-14 Thread Dan Austin

 Its illegal to interconnect it to the local pstn (from abroad).

 Dinesh.

 I still don't see how this would stop him from using no VOIP  
 protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,

 no VOIP, no bypassing the Indian telephone monopoly (assuming that  
 there is one).
Actually there are several.  It might not make much sense to you,
we either for that matter, but it is illegal.  To the point that
plugging in a system capable of VoIP, but not used for VoIP can get
you in trouble.

I needed to install a system in India and worked with the Telcos to
see if I could install a local-use only system.  We were offered the
option of buying a $24,000 license, which could be revoked at anytime
for any reason.  No one could tell us if we would need to renew
annually,
or if the license fees were refundable if the law changed.

So what makes sense to you and I had no bearing in India, and the 
rules don't look like they'll be changing anytime soon.

Dan
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RE: [Asterisk-Users] asterisk sample size adjustment

On Mon, 2005-11-14 at 09:11 -0800, Dan Austin wrote:
  hi,
  
  723 needs 30ms + 5ms forward. 729 needs 10ms + 5ms forward. The
  'standard' for 711 is actually 6ms (48 bytes). This would have to be
  done per channel (or per codec), but I am not sure wherever Asterisk
  allow per codec size or run's with one static size???
 
 There is a patch in Mantis, bugid 5162, I think, that allows for
 changing
 the packetization size.  It surrently works on  global/users/peers, but 
 is not per codec.
 
eeps that can break stuff or at least cause performance problems with
mixed codecs :(

At least its a start..  personal preference I dont like stuff hardcoded
unless it has to be, but that is just me.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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[Asterisk-Users] Polycom Buddy Feature

2005-11-14 Thread Araba, Michael









Yes. But will not monitor more than 7 of the phones. It lists
them in order of the last name entered into the directory not even in order of
the speed dial setting.



Any ideas? I have more 20 polycom 601/501 phones deployed



Michael










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[Asterisk-Users] Connecting analog lines to Asterisk for IP telephony device use

2005-11-14 Thread Wylie Swanson

I currently have broadvoice service for two lines and one analog line coming from Cox cable telephone. I would like to connect the single analog line to my asterisk server, and then use only IP-based telephony for all of my handsets. 
What is the best method for accomplishing connecting a single residential line to an Asterisk server? What is the least expensive method (if not the same thing)?Best regards,Wylie
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Re: [Asterisk-Users] newbie question regarding asterisk


On Nov 14, 2005, at 6:21 AM, Markos Paraskevopulos wrote:

Hello everyone,

I’m new to VoIP and despite a lot of reading, I’m kind of more  
confused than before.


I have following question – we currently have hardware Alcatel PBX  
and approx. 50 phones in the company. I was wondering if we would  
need to change the phone service provider, because they don’t  
provide VoIP services if we were about to switch to Asterisk  
instead of the Alcatel PBX?


Or can Asterisk maintain current functionality plus adding VoIP by  
simply switching the alcatel pbx for Asterisk server?


I hope I’m making at least a bit of sense.

 Thanks in advance for help

Confused

Markos

Markos,

The answer to your question is Maybe. It depends on how you connect  
your existing PBX to the PSTN, and it depends on what you want from  
your system.


Asterisk is completely capable of connecting to standard analog and  
digital (T1/E1/PRI) phone circuits. You do not need to use VOIP to  
connect Asterisk to the phone network. However, how you will go about  
doing this depends on your call volume and budget. How many incoming/ 
outgoing phone lines you have, how much long distance you dial, and  
local telco rates all play a part here.


The easiest way is to figure out how you connect the existing PBX,  
and then you can research to see if Asterisk will support that  
technology. (Chances are that it does). For example, if your Alacatel  
connects to the PSTN via a T1/E1 Circuit, then you could buy an T1/E1  
interface card from Digium or Sangoma and plug the T1/E1 right into  
your Asterisk server. If you have multiple analog POTS lines, then  
it's more complicated, but there are solutions for that, too (digium  
X100P, TDM400p, TDM2400p, various SIP gateways, multiple Sipura  
SPA-3000, etc...)


Then you might want to research your other options and make sure that  
you are using the most cost effective solution for your needs (This  
all depends on how you use the PSTN and what the local rates and  
availability are). The most basic knowledge you will need is the  
difference between a T1/E1 style connection and a regular analog POTS  
line. For example, if you have multiple analog lines, you might be  
able to save money by getting a full or fractional T1/E1.


If you're still completely confused and you don't have a lot of  
telecom knowledge, you might want to consider hiring a consultant to  
help you out.


Tom

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

Technology solutions for small and medium sized businesses.



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[Asterisk-Users] Media gateway recommendations?

2005-11-14 Thread Dustin Wenz

I am in the process of replacing a Zultys MX250 with an Asterisk PBX  
system. Right now, there is a PRI T1 line coming in from the phone  
company that plugs directly into the MX250. I believe Digium offers  
PCI cards that will provide this same functionality, but what I would  
really like is to have Asterisk physically abstracted from the phone  
company's interface. That would mean I'll need a standalone media  
gateway that serves SIP traffic to the LAN.


Zultys offers the MX25 which may do what I want, but I really want to  
avoid anything from Zultys at the moment. I think Cisco also offers a  
media gateway, but I haven't been able to find any case examples of  
how well it works with Asterisk in this sort of environment. Can  
anyone recommend a gateway that is reasonably priced and can be  
cleanly integrated with Asterisk?


Thanks!

- .Dustin Wenz
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RE: [Asterisk-Users] Anybody tried it from India ?.

On Mon, 2005-11-14 at 12:15 -0500, Dean Collins wrote:
 There are a number of asterisk implementations in India.

And the commercial ones are finally legal..  Right about the time they
arrest some guys running a VoIP shop they make it legal, all in the same
week.  Biggest concern india appears to have now are wiretapping (cant
tap an illegal network) and revenue (the bust that happened the gov was
complaining about all the lost revenue).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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RE: [Asterisk-Users] Anybody tried it from India ?.

On Mon, 2005-11-14 at 09:18 -0800, Dan Austin wrote:
  Its illegal to interconnect it to the local pstn (from abroad).
 
  Dinesh.
 
  I still don't see how this would stop him from using no VOIP  
  protocols and plugging it in to the PSTN. Just use Asterisk as a PBX,
 
  no VOIP, no bypassing the Indian telephone monopoly (assuming that  
  there is one).
 Actually there are several.  It might not make much sense to you,
 we either for that matter, but it is illegal.  To the point that
 plugging in a system capable of VoIP, but not used for VoIP can get
 you in trouble.

That law just changed, or so the news blubs indicated.  

The bust (same week as the changes I read about)
http://news.webindia123.com/news/showdetails.asp?id=160798cat=India

UPI reports that india changed stuff in their laws now
http://www.physorg.com/news8123.html

I didnt read the article in full before (and didnt read *that* article
specifically).  Its legal for outside india, if you want to call inside
india you have to use the regular telco stuff, but with the entry fees
being lowered that monopoly may be challenged.

For the most part it was quasi-legal, on the books as illegal but rarely
enforced, although for the one call shop it seems that it did get
enforced there ...

-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] NAT setup


On Nov 14, 2005, at 11:57 AM, Andre Courchesne - Consultant wrote:


Hi all,

 I am setting up a a proof on concept where a SIP phone sits on the  
internet and connects to a * behing a NAT.


 Right now the SIP phone connects to the * box just fine, I can  
dial and I see the commands being executed on the * box, but I  
don't have any audio on the SIP phone. Any idas/pointers?


I would recommend that you do a little research on google, voip- 
info.org, and the list archives.


To connect to an Asterisk box that sits behind NAT, you need to  
forward ports 5060 and 1-2 too the asterisk box, and you need  
to configure the externip, localnet, and nat variables in sip.conf.  
audio problems are almost always due to the RTP stream (ports  
1-2) not being forwarded properly, either due to the port  
forwarding setup or the sip.conf settings.


Tom

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Re: [Asterisk-Users] Fritz card usb v2.1 - Capi installation problem

2005-11-14 Thread Armin Schindler

If 'capiinfo' does not work, chan_capi will fail too.

Do you have the node /dev/capi20 with correct permissions?

Armin

On Mon, 14 Nov 2005, Amaury BOSSE wrote:
 Hi all,
 
  
 
 I am trying to install an AVM Fritz card USB v2.1 on my Asterisk Box.
 
 I am using Debian Sarge with 2.6.8 kernel.
 
 I have compiled capi last drivers (fcusb2-suse93-3.11-07.tar.gz) and have
 copied fcusb.ko to /lib/modules/2.6.8/extra/.
 
  
 
 All modules seems loaded (capi, capifs, kernelcapi, fcusb,.)
 
 Capiinfo fails and returns : capi not installed - No such device or
 address (6)
 
  
 
 I have tried to install chan_capi-cm-0.6.1.tar.gz but Asterisk no longer
 starts.
 
 /var/log/asterisk/messages returns :
 
 Nov 14 16:40:51 WARNING[4005]: CAPI not installed, CAPI disabled!
 
 Nov 14 16:40:51 WARNING[4005]: chan_capi.so: load_module failed, returning
 -1
 
 Nov 14 16:40:51 WARNING[4005]: Loading module chan_capi.so failed!
 
  
 
 I have tried to find out a solution from the web but without results.
 
 Does someone know where the problem is from?
 
  
 
 Thanks for your help
 
 Amaury
 
 
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Re: [Asterisk-Users] Connecting analog lines to Asterisk for IP telephony device use


On Nov 14, 2005, at 12:24 PM, Wylie Swanson wrote:

I currently have broadvoice service for two lines and one analog  
line coming from Cox cable telephone.  I would like to connect the  
single analog line to my asterisk server, and then use only IP- 
based telephony for all of my handsets.


What is the best method for accomplishing connecting a single  
residential line to an Asterisk server?  What is the least  
expensive method (if not the same thing)?


Best regards,

Wylie


The least expensive, I think, is a cloned X100P card from eBay.

The way *I* would recommend is to use a Sipura SPA-3000.

Tom

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Cascade Link Systems
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(603) 375-1414

Technology solutions for small and medium sized businesses.



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Re: [Asterisk-Users] Connecting analog lines to Asterisk for IP telephony device use

2005-11-14 Thread Cory Andrews

You need a Digium TDM01B which will allow you to connect your single 
analog (FXO) line to your Asterisk server.  From there, you can go 
Ethernet out of your Asterisk box, through a switch, to your IP endpoints.


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



Wylie Swanson wrote:
I currently have broadvoice service for two lines and one analog line 
coming from Cox cable telephone.  I would like to connect the single 
analog line to my asterisk server, and then use only IP-based 
telephony for all of my handsets. 

What is the best method for accomplishing connecting a single 
residential line to an Asterisk server?  What is the least expensive 
method (if not the same thing)?


Best regards,

Wylie


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Re: [Asterisk-Users] NAT setup

2005-11-14 Thread Carlos Chavez





On Mon, 2005-11-14 at 11:57 -0500, Andre Courchesne - Consultant wrote:


Hi all,

  I am setting up a a proof on concept where a SIP phone sits on the 
internet and connects to a * behing a NAT.

  Right now the SIP phone connects to the * box just fine, I can dial 
and I see the commands being executed on the * box, but I don't have any 
audio on the SIP phone. Any idas/pointers?



 That usually means that you have not forwarded the RTP ports. By default Asterisk uses ports 1 - 2 for RTP on UDP so you need to tell your NAT device to forward those ports to your * box.





-- 
Carlos Chavez
Director de Tecnologa
Telecomunicaciones Abiertas de Mxico S.A. de C.V.
Tel: +52-55-91169161 Ext 2001








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Re: [Asterisk-Users] How do I know if I have CRC-CCITT (README.Linux26)

2005-11-14 Thread Mojo with Horan Company, LLC

try typing modprobe crt-ccitt before the commands Chuck wrote, or 
alternatively, if you've got your slocate database up to date, you could 
try locate crc-ccitt.ko


ram wrote:

Hi
 
i have installed FC4 with select everything

with your command i dont find any thing
results are null
thats means its not installed ??
 
ram


 
On 11/14/05, *Saul Diaz* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Chuck Bunn wrote:

You have

lsmod  | egrep crc_ccitt

check in your kernel modules looking for crc_ccitt

but FC4 comes with that

regards
Saul

  Hi,
 
  I have downloaded the latest release candidate v1-2-0-rc2 and I was
  checking the readme's in the zaptel directory and I came across a
  requirement I have not seen before. In the readme it says that for
  Linux 2.6 kernel you will need to have the 'CRC-CITT' functions
  compiled with the kernel. I am using Fedora 4 kernel version
  2.6.13-1.1532.FC4. How can I verify that I have these functions?
 
  Thanks
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] Media gateway recommendations?

2005-11-14 Thread Marc Rys

Look into the Lucent Max Tnt.  It may be over kill for your application, but 
people say they work flawlessly and will increase the stability of your 
asterisk PBX.  I'm in the process of setting one up right now.

Good Luck

Marc

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wenz
Sent: Monday, November 14, 2005 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Media gateway recommendations?

I am in the process of replacing a Zultys MX250 with an Asterisk PBX  
system. Right now, there is a PRI T1 line coming in from the phone  
company that plugs directly into the MX250. I believe Digium offers  
PCI cards that will provide this same functionality, but what I would  
really like is to have Asterisk physically abstracted from the phone  
company's interface. That would mean I'll need a standalone media  
gateway that serves SIP traffic to the LAN.

Zultys offers the MX25 which may do what I want, but I really want to  
avoid anything from Zultys at the moment. I think Cisco also offers a  
media gateway, but I haven't been able to find any case examples of  
how well it works with Asterisk in this sort of environment. Can  
anyone recommend a gateway that is reasonably priced and can be  
cleanly integrated with Asterisk?

Thanks!

 - .Dustin Wenz
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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

how can i check how many g729 are being used right now?On 11/13/05, Gentian Bajraktari [EMAIL PROTECTED]
 wrote:Yes.- Original Message -From: Angelito Manansala 
[EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com
Sent: Sunday, November 13, 2005 1:23 PMSubject: Re: [Asterisk-Users] How to check how many G729 codeclicenseinstalled
g723 gsmulawalawg726
adpcmslin lpc10g729 speexilbc
g723 -
- -
- -
- -
- -
- -gsm
- -
3 3
4 3
2 9
- - 131
ulaw -
5 -
1 3
2 1
8 - - 130
alaw -
5 1
- 3
2 1
8 - - 130
g726 -
6 3
3 -
3 2
9 - - 131adpcm
- 5
2 2
3 -
1 8
- - 130
slin -
4 1
1 2
1 -
7 - - 129lpc10
- 8
5 5
6 5
4 -
- - 133
g729 -
- -
- -
- -
- -
- -speex
- -
- -
- -
- -
- - -
ilbc -
9 6
6 7
6
512
- - - this means i have no g729 codec installed.. thanks guys! :p On 11/13/05, Gentian Bajraktari 
[EMAIL PROTECTED] wrote: Do: *CLI show translations If you see - (lines) on the G729 row/columns than you do not have any G729 support.
 RG. - Original Message - From: Sahil Gupta [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com Sent: Sunday, November 13, 2005 1:03 PM Subject: Re: [Asterisk-Users] How to check how many G729 codec
 licenseinstalled  Right :)   Regards,Sahil Gupta  VoiceValley 
  On Sun, 13 Nov 2005, Angelito Manansala wrote:   *CLI show g729  No such command 'show g729' (type 'help' for help)   this means i have no g729 codec installed, right?
   On 11/13/05, Zafer Khodr [EMAIL PROTECTED] wrote:  That's easy...  Just go into asterisk cli and type show g729
  It will tell you how many are active and how many you have in totalRegards  Zafer 
  -Original Message-  From: [EMAIL PROTECTED]  [mailto:
[EMAIL PROTECTED]] On Behalf Of  Angelito  Manansala  Sent: Sunday, 13 November 2005 10:31 PM  To: 
asterisk-users@lists.digium.com  Subject: [Asterisk-Users] How to check how many G729 codec license  installed 
  Guys, is the any CLI commands or info files where you can check how  many g729 codec  license installed.  
  Regards,  Lito  ___  --Bandwidth and Colocation sponsored by 
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 http://lists.digium.com/mailman/listinfo/asterisk-users   
  --  Best Regards,  Angelito Manansala  www.voicefidelity.net  Mobile: +639175425807
  DID: (+63) 44 7906770  msn: [EMAIL PROTECTED]  skype: bulcrack  ___
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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net
 Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by 
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Mark Quitoriano, CCNAhttp://www.atamanetworks.com
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[Asterisk-Users] newbie question regarding asterisk

2005-11-14 Thread Markos Paraskevopulos









Hello everyone,

Im new to VoIP and despite a lot of reading, Im
kind of more confused than before.

I have following question  we currently have
hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if
we would need to change the phone service provider, because they dont
provide VoIP services if we were about to switch to Asterisk instead of the
Alcatel PBX?

Or can Asterisk maintain current functionality plus
adding VoIP by simply switching the alcatel pbx for Asterisk server?

I hope Im making at least a bit of sense.



Thanks in advance for help



Confused



Markos






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Re: [Asterisk-Users] How to check how many G729 codec licenseinstalled

2005-11-14 Thread Gentian Bajraktari




Dependent on what Channel you are 
using..
If you are using SIP then do:

*CLI sip show channels


Peer 
User/ANR Call ID Seq 
(Tx/Rx) Format Last Msg



Where Format will show the current codecs used at 
that time..

Rg,

Gentian


  - Original Message - 
  From: 
  Mark 
  Quitoriano 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Monday, November 14, 2005 12:13 
  PM
  Subject: Re: [Asterisk-Users] How to 
  check how many G729 codec licenseinstalled
  how can i check how many g729 are being used right now?
  On 11/13/05, Gentian 
  Bajraktari [EMAIL PROTECTED]  
  wrote:
  Yes.- 
Original Message -From: "Angelito Manansala"  [EMAIL PROTECTED]To: 
"Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.com 
Sent: Sunday, November 13, 2005 1:23 PMSubject: Re: 
[Asterisk-Users] How to check how many G729 
codeclicenseinstalled 
g723 gsmulawalawg726 
adpcmslin lpc10g729 
speexilbc g723 
- - 
- - 
- - 
- - 
- - 
-gsm 
- - 
3 3 
4 3 
2 9 
- - 131 
ulaw - 
5 - 
1 3 
2 1 
8 - - 
130 alaw 
- 5 
1 - 
3 2 
1 8 
- - 130 
g726 - 
6 3 
3 - 
3 2 
9 - - 
131adpcm 
- 5 
2 2 
3 - 
1 8 
- - 130 
slin - 
4 1 
1 2 
1 - 
7 - - 
129lpc10 
- 8 
5 5 
6 5 
4 - 
- - 133 
g729 - 
- - 
- - 
- - 
- - 
- -speex 
- - 
- - 
- - 
- - 
- - - 
ilbc - 
9 6 
6 7 
6 
512 
- - 
- this means i have no g729 codec 
installed.. thanks guys! 
:p On 11/13/05, Gentian Bajraktari  [EMAIL PROTECTED] 
wrote: Do: *CLI show 
translations If you see - (lines) on the G729 
row/columns than you do not have any G729 support. 
 RG. - 
Original Message - From: "Sahil Gupta" [EMAIL PROTECTED] 
To: "Asterisk Users Mailing List - Non-Commercial Discussion"  
asterisk-users@lists.digium.com 
Sent: Sunday, November 13, 2005 1:03 PM Subject: Re: 
[Asterisk-Users] How to check how many G729 codec  
licenseinstalled  Right 
:)   Regards,  
  Sahil Gupta  VoiceValley 
   On Sun, 13 Nov 2005, Angelito Manansala 
wrote:   *CLI show g729 
 No such command 'show g729' (type 'help' for help) 
  this means i have no g729 codec installed, 
right?On 11/13/05, Zafer Khodr 
[EMAIL PROTECTED] 
wrote:  That's easy...  Just 
go into asterisk cli and type" show g729" 
  It will tell you how many are active and how many 
you have in total  
  Regards  
Zafer-Original 
Message-  From: [EMAIL PROTECTED] 
 [mailto: 
[EMAIL PROTECTED]] On Behalf Of 
 Angelito  Manansala 
 Sent: Sunday, 13 November 2005 10:31 PM 
 To: asterisk-users@lists.digium.com 
 Subject: [Asterisk-Users] How to check how many G729 codec 
license  installed  
  Guys, is the any CLI commands or info files where 
you can check how  many g729 codec 
 license installed.  
   Regards,  
Lito  
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--  Best Regards,  Angelito 
Manansala  www.voicefidelity.net 
 Mobile: +639175425807   DID: (+63) 44 
7906770  msn: [EMAIL PROTECTED] 
 skype: bulcrack  
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Re: [Asterisk-Users] iax-qos-openbsd...

2005-11-14 Thread Nurdin

Have you try Arbitrator Open source based on Linux, there is called 
ArbiQos which optimum for Voip and Video stream, that has priority the 
bandwidth for voip and it can gone when no others voip or video stream used



Francois Meehan wrote:

Hi all,

We have an asterisk server inside a network using an iax provider. The
firewall is based on Openbsd, and we would like to use PF's QOS
capabilities to ensure optimum quality.

We need to provide good throughput for other applications, so we need to
use scheme that borrows bandwith, that is when there is no VOIP
communication, the whole upload capability of our link can be use.

We have tried all kind of combinations but could not come up with a
satisfactory solution.

As anyone faced a similar configuration, if so how did you deal with that?

Regards,

Francois
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Re: [Asterisk-Users] Asterisk realtime extensions context inclusion

2005-11-14 Thread bbench

Don't know will this do, but a simple comparison 
may give you a hint:
1.2-rc2 extentions.conf
[default]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
;switch = Realtime/[EMAIL PROTECTED]

1.0.9 extentions.conf
[default]
include = astcc
include = internal
[astcc]
exten = _011N.,1,Set(CALLERID(name)=${CALLERIDNAME} - ${CALLERIDNUM})
exten = _011N.,2,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN},4)
exten = _011N.,3,Hangup
[internal]
exten = _4XXX,1,Dial(sip/${EXTEN},20,r)
or_whatever = _way, you, use
Both do the same job. See how simple is with realtime?
You could do also:
[default]
switch = Realtime/[EMAIL PROTECTED]
switch = Realtime/[EMAIL PROTECTED]
[default_vonage]
switch = Realtime/[EMAIL PROTECTED]
[default_sipgate]
switch = Realtime/[EMAIL PROTECTED]
Hope it helps?
benchev

On Monday 14 November 2005 09:44, Daniel Clark wrote:
 Thanks for the reply, it's an approach I didn't think of to simply include
 the information from the other contexts into where I would be including
 from. In most cases that would work, but not in my case. Each user of my
 system will be able to place outgoing calls using their own sip connection
 (as in one they create with sipgate or vonage etc). To ensure that each
 user can dial out with their own sip connection and nobody else's they are
 each getting their own context and that context is the only place in the
 dialplan to dial that particular external sip connection. For a small
 amount of users it's possible to include all the information in each
 context, however I'm dealing with 15,000 users and would like a database
 small enough to fit on the hard disk!



 Would it not be possible to do something with the Goto app? In each persons
 dialplan I can have an extension to catch internal numbers and then forward
 to another context using exten = 1,1,Goto(context2) or something like
 that?



 I have to stick with the database option as there are other applications
 that need quick access to the information it holds. It's not really
 possible to generate the flat file for all the contexts when at some times
 that would mean generating the file over 1,000 a day and reloads of the
 database each time. If I can stick with the realtime database in any way, I
 would much prefer to.



   _

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of snacktime
 Sent: 14 November 2005 07:24
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk realtime extensions context
 inclusion





 On 11/13/05, Daniel Clark [EMAIL PROTECTED] wrote:

 Hi



 I'm using asterisk realtime to control all of my extensions. As part of
 this I need to be able to dynamically create new contexts and extensions.
 The new contexts I create will also include existing contexts. Does anybody
 know the how to specify context inclusion for asterisk realtime as the
 database only has colums for id, context, exten, priority, app and appdata.


 You can't.  Since those other contexts are in the database, why not just
 select them and then insert them into the newly created context?

 Or better yet dump realtime and generate extensions.conf from your own
 database schema.   You could even use the realtime schema with just a
 couple of extra fields for things like include files, that way you dont'
 have to throw away the work you have already done.

 Asterisk doesn't handle database failures very well.  Maybe it's been fixed
 now, but for instance a dialplan reload used to wipe out your whole
 dialplan if the database was down instead of just skipping the reload.  I
 spent quite a bit of time writing an application for ARA at one point, only
 to toss it all out after seeing how it actually worked.  I still think it's
 a good idea,  and I don't mean to disparage those who put all the work into
 it, but it's implementation leaves something to be desired.

 Chris
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RE: [Asterisk-Users] ISDN card required

2005-11-14 Thread Mark Elkins

So far - the 4-port ISDN HFC chipset cards from Junghanns.net works well
and is half the price of a 4-port Eicon card.


On Mon, 2005-11-14 at 10:07 +, David Waugh wrote:
 Hi Lee,
  
 I use a Diva Server card here with Asterisk using Chan_capi.
 The basic BRI card has one BRI port. They also have a model with 4
 port BRI model. You can mix and match Diva Server card too, so as your
 needs expand you can add more cards to your server.
  
 Further information can be found on the Eicon website:
  
 http://www.eicon.com/worldwide/solutions/Diva_Server_and_Asterisk
  
 and
 http://www.eicon.com/worldwide/products/MediaGateways/all-in-one.htm
  
 Thanks
 David
  
  
  
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 Lee Archer
 Sent: 14 November 2005 09:32
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] ISDN card required
 
 
 
 Can anyone point me in the direction of a quality, works with
 Asterisk, BRI card.  I need minimum 2 port/4 channel. 
 
 Regards 
 
 Lee 
 
 ###
 
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RE: [Asterisk-Users] Asterisk Installation exits with following error ***

2005-11-14 Thread Zeeshan

How do I install curl?

Zeeshan A Zakaria


-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED] 
Sent: Sunday, November 13, 2005 10:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Installation exits with following
error


 After running make clean; make install, asterisk starts installing
itself but then 
terminates after some time with the following error:
 
  
 
 /usr/bin/ld: cannot find -lidn
 
 collect 2: Id returned 1 exit status
 
 make[1]: *** [app_curl.so] Error 1
 
 make[1]: Leaving directory '/usr/src/asterisk/apps'
 
 make: *** [subdirs] Error 1
 
  
 
  
 
 On typing asterisk at command prompt doesn't start asterisk, which
mean it didn't 
install successfully. I never had this problem
 previousely, what am I mississing this time. Did I miss something when
installing 
Fedora Core 3?
 

The message is suggesting you don't have 'curl' installed. (I had the
same issue some time ago with fc3.)

Download curl (and if I'm not mistaken, you need the curl source code
installed as well).

Then do another 'make clean' followed by 'make install'.


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Re: [Asterisk-Users] iax-qos-openbsd...

2005-11-14 Thread Bartosz Jozwiak





Have you try Arbitrator Open source based on Linux, there is called 
ArbiQos which optimum for Voip and Video stream, that has priority the 
bandwidth for voip and it can gone when no others voip or video stream 
used



Francois Meehan wrote:

Hi all,

We have an asterisk server inside a network using an iax provider. The
firewall is based on Openbsd, and we would like to use PF's QOS
capabilities to ensure optimum quality.

We need to provide good throughput for other applications, so we need to
use scheme that borrows bandwith, that is when there is no VOIP
communication, the whole upload capability of our link can be use.

We have tried all kind of combinations but could not come up with a
satisfactory solution.

As anyone faced a similar configuration, if so how did you deal with 
that?


You can use HTB if I am not mistaken ...

B. 


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RE: [Asterisk-Users] Anybody tried it from India ?.

2005-11-14 Thread Mark Elkins

I can not see that its illegal to have Asterisk in India. The TDM400P
card should work fine - but it may not be approved to be interconnected
to the phone system. (This never stopped me doing similar things).

I'm assuming that its possible to connect a 2-wire phone to the Indian
phone system - ie - if you have ever bought a 2-wire phone from the USA
and got it to work - then there should be no problem. In the UK - BT use
a 3-wire system, the extra wire for ringing a bell... but actually
provide 2-wire to the house. People seem to have little difficulty with
the TDM400 there. I've had no problems all over Africa - you should be
fine.

Asterisk makes a great (cost wise) and highly functional PABX
replacement. This in itself is reason to install Asterisk.

The fact that it does VoIP as well is an additional bonus - just don't
get caught using it?

Up until the beginning of this year, VoIP was illegal in South Africa -
never stopped most people. It is possible for telco's to monitor and
even recognise and record 'voice' on the internet - but they usually
look for common Codecs (u-law, a-law) and probably have better things to
do. 


On Mon, 2005-11-14 at 15:50 +0800, Dinesh wrote:
 
 Its illegal to interconnect it to the local pstn (from abroad). 
 
 Dinesh.
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes
 Sent: Monday, November 14, 2005 1:43 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Anybody tried it from India ?.
 
 On Nov 14, 2005, at 12:37 AM, ram wrote:
 
  Hi
 
  its not legal in india
  connecting to PSTN to VOIP
 
  ram
 
 Asterisk doesn't necessarily mean VOIP. He could set it up using ZAP  
 channels only and not have any VOIP in use at all.
 
 Tom
 
 
 Cascade Link Systems
 www.cascadelinksystems.com
 (603) 375-1414
 
 Intelligent technology solutions for small businesses.
 
 
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 DISCLAIMER:
 This email is confidential and may be privileged. If you are not the intended 
 recipient, please delete it and notify us immediately. Please do not copy or 
 use it for any purpose, or disclose its contents to any other person as it 
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 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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Re: [Asterisk-Users] newbie question regarding asterisk

On Mon, 2005-11-14 at 12:21 +0100, Markos Paraskevopulos wrote:
 Hello everyone,
 
 I’m new to VoIP and despite a lot of reading, I’m kind of more
 confused than before.
 
I had an asterisk system up and running then read some dox and becuase
what I read at that time wasnt well written it has that effect :)

asteriskdocs.org is pretty good, and the oreilly book asterisk and the
future of telephony (pdf is available at asteriskdocs.org) is a good
read, and only takes about 1 night to read everything except the
appendices.


 I have following question – we currently have hardware Alcatel PBX and
 approx. 50 phones in the company. I was wondering if we would need to
 change the phone service provider, because they don’t provide VoIP
 services if we were about to switch to Asterisk instead of the Alcatel
 PBX?
 

Asterisk does more than VoIP.  It can speak analog (both fxs and fxo -
or act like a phone company (fxs) or act like a phone (fxo)), it can
also do digital trunks (t1/e1/j1 - ds3 soon alledgly).  While it can
replace a pbx it can also provide a T1 to a pbx.  It can talk to the
phone company via VoIP or whatever circuits you already have.

You dont *have* to switch phone companies if you dont want to, and it
doesnt always make sense to switch.  


 Or can Asterisk maintain current functionality plus adding VoIP by
 simply switching the alcatel pbx for Asterisk server?
 
If you want to add VoIP you can do this more gradually if you dont have
the budget to totally replace 100%.  Asterisk can feed your current pbx
with phone service, where it interconnects to can be either VoIP or PSTN
or both.  Eventually you can migrate off what you already have.

If however you have the budget to replace every phone on the desktop (or
get appropriate interface equipment so the phones can speak to asterisk)
then asterisk should be able to maintain current functionality plus
adding anything that it does that you dont have (ie VoIP).  


 I hope I’m making at least a bit of sense.
 
I hope my answer makes sense..  

 
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/  Sacramento Asterisk Users Group


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RE: [Asterisk-Users] Asterisk realtime extensions context inclusion

2005-11-14 Thread Sergey Okhapkin

That's what macro is useful for. Don't include these common contexts, but convert them to macros and call these macros from user's dialplan. Macros in asterisk dialplan are close to subroutines rather than to C-style #define.

On Mon, 2005-11-14 at 07:44 +, Daniel Clark wrote:

Thanks for the reply, its an approach I didnt think of to simply include the information from the other contexts into where I would be including from. In most cases that would work, but not in my case. Each user of my system will be able to place outgoing calls using their own sip connection (as in one they create with sipgate or vonage etc). To ensure that each user can dial out with their own sip connection and nobody elses they are each getting their own context and that context is the only place in the dialplan to dial that particular external sip connection. For a small amount of users its possible to include all the information in each context, however Im dealing with 15,000 users and would like a database small enough to fit on the hard disk!

Would it not be possible to do something with the Goto app? In each persons dialplan I can have an extension to catch internal numbers and then forward to another context using exten = 1,1,Goto(context2) or something like that?

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RE: [Asterisk-Users] Sipura SPA-2002 Double Ring

2005-11-14 Thread Trevor G. Hammonds

Rich Adamson wrote on Sunday, 13 November 2005 7:36 PM:

 I recently implemented a Sipura SPA-2002 with one of my Asterisk
 installations.  On internal calls, the SPA generates ringtone as
 expected. However, when I dial out via my IAX-based service
 provider, I hear 
 both the telco-generated ringtone as well as the SPA-generated
 ringtone.  Sometimes, the SPA continues to generate the ringtone even
 after the call has been answered.
 
 I don't have a spa-2002, but do use a spa3k. I doubt very much the
 sipura device is actually providing ringback tone, and I don't recall
 any parameters that would enable/disable such an item. (The Admin
 manual does not mention it either.)   
 
 You might check your extensions.conf entry for dialing your provider
 to see if you have an r in that line. If so, remove it. 

The SPA-2002 is definitely generating the additional ringback.  I verified
this by temporarily changing the frequency of the ringback in the SPA's
Regional settings.  

I also verified that I am not using the r option in the Dial command.  If
I were, however, only the Asterisk-generated ringback would be heard, and
then only until the call supervised (i.e. I would not be hearing two
distinct ring signals, and the ringback would not occasionally persist for
the duration of a call while still hearing the called party).  

This problem is present only with the SPA-2002, and none of the other SIP
devices connected to this Asterisk server.  I have also tried making
outbound calls via different service providers, all with the same results.  

Thanks again.  

Sincerely,
Trevor Hammonds



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[Asterisk-Users] Snom clients deregistering