date:20050117

Am Montag, 17. Januar 2005 09:14 schrieb Vamsi Pottangi:
 Hi,
 Is there a way of adding SIP clients using AGI ? I see that, only
 extensions can be added using the AGI.
 If not AGI, is there any other way of adding SIP clients other than
 editing siop.conf manually ?
 Thanks,
 ~Vamsi

You can of course write an AGI script/program that adds entries to sip.conf. 
As far as I know, there is no explicit Asterisk command to add new SIP peers.

Robert Spielmann
-
TAL.DE  Klaus Internet Service GmbH [EMAIL PROTECTED]
Robertstr. 6        *      D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364  *  Fax +49 (0) 202 / 495-399

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Re: [Asterisk-Users] No more loading asterisk...

Am Samstag, 15. Januar 2005 22:39 schrieb Scheda:
 Hey, whenever I try to load, I get these errors

 Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to
 bind to 0.0.0.0 port 4569: Address already in use
 Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource:
 chan_iax2.so: load_module failed, returning -1
   == Manager unregistered action IAXpeers
   == Unregistered channel type 'IAX2'
 Jan 15 16:37:24 WARNING[7573]: loader.c:440 load_modules: Loading
 module chan_iax2.so failed!


 A friend reccomended I used Apt-Get to install it, so I tryed to do
 that to overwrite it, and even after a full recompile I get these
 errors... I don't quite know what to do.

If you (re-)installed Asterisk via apt-get, it is quite certainly started 
automatically at boot time (via a script in /etc/init.d). If so, Asterisk is 
already running when you try to start it manually, and hence port 4569 is 
already open and cannot be used by the newly started asterisk.

Cheers
Robert Spielmann
-
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Robertstr. 6        *      D-42107 Wuppertal, Germany
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[Asterisk-Users] Euro ISDN and Caller ID (Sweden)

2005-01-17 Thread Daniel Nyström

Do anyone have experiences with Euro ISDN in Sweden?
Does CallerID work properly? Both in and out.
Do anyone know of a reseller for Digium cards and/or CarrierAccess Adit 600 in 
Sweden or Europe (EU)?

Thanks!

BR
Daniel Nyström
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[Asterisk-Users] AGI / Sockets

Hi,

what happens if the dialplan contains something like

exten = s,1,AGI(agi://10.0.0.1)
exten = s,2,Dial(SIP/phone1|20|tr)

etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my 
test cases, I always got a hangup and no further processing of the dialplan. 
Any hints? ( the call mustn't go into Nirvana if the AGI server isn't 
available!)

Thanks for any help
Robert Spielmann
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[Asterisk-Users] ASTCC single stage + no access number + auth using sip username and password

2005-01-17 Thread Krystian Filiks









Hi All,



Im new to * 

I wonder if anyone have an idea how to make the following
with ASTCC.



I would like to have all SIP phones to work on prepaid basis
and without need to dial any access number, instead I would like to use the
phone as normal dialing only the destination number, for example 00464090510.



Then I would like to authenticate the call using the SIP
Username and Secret check for balance, set timeout, inform the user about the
time limit of the call and setup the call to an H323 GW.



Once the call is finished I would like to have the balance
shown in the display by sending a sip message to the phone(if possible
otherwise not important).



I was searching the net for info about ASTCC but there is
not much that I found which could help me.



Thanks In Advance

KF










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[Asterisk-Users] Can I get info about email addresses from voicemail.conf in dialplan or variables ?

2005-01-17 Thread Robert Rozman

Hi,

I'd like to setup automatic recording of channels and send wav files via
email to extension user (to same email address as in voicemail.conf). Can I
access those addresses from dialplan or AGI ?

Regards,

Rob.

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[Asterisk-Users] Does Asterisk do that?

2005-01-17 Thread Alberto Martnez

Hello.

I have just arrived to Asterisk. I would like to know if Asterisk can
perform some functionalities I am looking for.

I want to allow voip over sip to some users. All of them must have
their own user name and password to login to Asterisk so only allowed
users can login. All calls started by users have to be redirected to
one account at our voip provider. I think those functionalities can be
covered with Asterisk but I would like to confirm it.

We are interested too in having a real-time control of callings,
knowing which calls are actives, its time and other datas. Is there
anyway to have this information with Asterisk? We are interested y
having it by consulting some files, for example, with a little piece
of software we develop.

We need to be able to finish a call by software, when certains
conditions are accomplished. Can we do that with Asterisk?

Thank you very much for your help.

Regards,
Alberto

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Re: [Asterisk-Users] Does Asterisk do that?

Am Montag, 17. Januar 2005 11:26 schrieb Alberto Martínez:
 Hello.

 I have just arrived to Asterisk. I would like to know if Asterisk can
 perform some functionalities I am looking for.

 I want to allow voip over sip to some users. All of them must have
 their own user name and password to login to Asterisk so only allowed
 users can login. All calls started by users have to be redirected to
 one account at our voip provider. I think those functionalities can be
 covered with Asterisk but I would like to confirm it.

Yes, that's possible.

 We are interested too in having a real-time control of callings,
 knowing which calls are actives, its time and other datas. Is there
 anyway to have this information with Asterisk? We are interested y
 having it by consulting some files, for example, with a little piece
 of software we develop.

 We need to be able to finish a call by software, when certains
 conditions are accomplished. Can we do that with Asterisk?

For the above 2 cases, look at the Asterisk Manager API, described for example 
at http://www.voip-info.org/wiki-Asterisk+Manager+API

Have fun and good luck!

Robert Spielmann
-
TAL.DE  Klaus Internet Service GmbH [EMAIL PROTECTED]
Robertstr. 6        *      D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364  *  Fax +49 (0) 202 / 495-399

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Re: [Asterisk-Users] H323 Softphone for iPAQ

2005-01-17 Thread Michael Manousos

Also the following has worked great for me:
http://www.wifive.net/introduction.asp
Michael
Radovan Mihalik wrote:
http://www.sjlabs.com/sjp.html
 
SJphoneR is a VOIP softphone that allows you to speak with any PC, PDA,
stand-alone IP-phone and with any legacy wired or mobile phone (using
your VOIP gateway or purchasing service from Internet Telephony Service
Provider). It supports both SIP and H.323 standards and is fully
interoperable with most major IP-telephony vendors and ITSP.
 
I'm just about to try it my self ;)
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab
Sent: Sunday, January 16, 2005 8:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] H323 Softphone for iPAQ
 
Hi list,
 
I was just wondering, is there any H.323 soft-phone that can be
installed on a pocket PC (iPAQ). 
 
Walid
 
 



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Re: [Asterisk-Users] chan_capi-0.3.5 error 127

2005-01-17 Thread Vincent Guidoux

after # vi Makefile

and changes in coment the ligne with gcc
i have the same error before

  In file included from /usr/include/linux/kernelcapi.h:13,
  from /usr/include/linux/capi.h:18,
  from chan_capi.c:35:
  /usr/include/linux/list.h:604:2: warning: #warning don't include
  kernel headers in userspace
  chan_capi.c: In function 'capi_new':
  chan_capi.c:1076: error: structure has no member named 'cid'
  chan_capi.c:1077: error: structure has no member named 'cid'
  chan_capi.c: In function 'capi_handle_dtmf_fax':
  chan_capi.c:1189: error: structure has no member named 'cid'
  chan_capi.c: In function 'pipe_msg':
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c: In function 'load_module':
  chan_capi.c:2843: warning: passing arg 4 of 'ast_channel_register'
  from incompatible poiner tpe
  make: *** [chan_capi.o] Error 1


On Wed, 12 Jan 2005 18:24:02 +0100, adria vidal [EMAIL PROTECTED] wrote:
 
 El 12/01/2005, a las 15:36, Vincent Guidoux escribi:
 
   
 
  Hi,
 
  I have a problem for install chan_capi
 
   
  My pc: Suse 9.1, with asterisk current stable en cvs
   
  I have download
  http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz
   
  And the path from
  http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
   
  And patch the chan_capi
   
  # cd chan_capi-0.3.5
  # patch p1  chan_capi-0.3.5-patch.diff
  # make/for install
   
  My error :
   
  Make: gcc-2.95 :Command not found
  Make: *** [chan_capi.o] Error 127
   
  If
   
  # make CC=gcc-3.3.3/because I have gcc-3.3.3
   
  Error:
   
  Make: gcc-2.95 :Command not found
  Make: *** [chan_capi.o] Error 127
   
  If
   
  # make CC=gcc
  In file included from /usr/include/linux/kernelcapi.h:13,
  from /usr/include/linux/capi.h:18,
  from chan_capi.c:35:
  /usr/include/linux/list.h:604:2: warning: #warning don't include
  kernel headers in userspace
  chan_capi.c: In function 'capi_new':
  chan_capi.c:1076: error: structure has no member named 'cid'
  chan_capi.c:1077: error: structure has no member named 'cid'
  chan_capi.c: In function 'capi_handle_dtmf_fax':
  chan_capi.c:1189: error: structure has no member named 'cid'
  chan_capi.c: In function 'pipe_msg':
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c:1764: error: structure has no member named 'cid'
  chan_capi.c: In function 'load_module':
  chan_capi.c:2843: warning: passing arg 4 of 'ast_channel_register'
  from incompatible poiner tpe
  make: *** [chan_capi.o] Error 1
   
   
  I don't know if the problem from gcc or oder, if you can help me,
  thanks soo much!
 
 
 coment lines into makefile looking for an specific gcc, and it will
 compile fine.
 
 Adri Vidal
 
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Re: [Asterisk-Users] Does Asterisk do that?

2005-01-17 Thread Jens Kbler

Am Montag 17 Januar 2005 11:26 schrieb Alberto Martínez:
 Hello.

 I have just arrived to Asterisk. I would like to know if Asterisk can
 perform some functionalities I am looking for.

 I want to allow voip over sip to some users. All of them must have
 their own user name and password to login to Asterisk so only allowed
 users can login. All calls started by users have to be redirected to
 one account at our voip provider. I think those functionalities can be
 covered with Asterisk but I would like to confirm it.
Confirmation


 We are interested too in having a real-time control of callings,
 knowing which calls are actives, its time and other datas. Is there
 anyway to have this information with Asterisk? We are interested y
 having it by consulting some files, for example, with a little piece
 of software we develop.
This information is available inside of asterisk console but maybe a fancy AGI 
script can export this information.

 We need to be able to finish a call by software, when certains
 conditions are accomplished. Can we do that with Asterisk?
Depends on the conditions. If you have nothing super special, * will handle it 
fine.

Jens
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Re: [Asterisk-Users] chan_capi-0.3.5 error 127

2005-01-17 Thread Frank Sautter

hi vincent,
Vincent Guidoux schrieb:
I have a problem for install chan_capi
My pc: Suse 9.1, with asterisk current stable en cvs
And patch the chan_capi
 chan_capi.c:1076: error: structure has no member named cid
as you are writing and apparent to the error message you are posting, 
you are using a stable 1.0.x version of asterisk.
therefore you don't need to apply my patch, which is only for the 
HEAD-cvs version of asterisk.
the version 0.3.5 from junghanns.net will (hopefuly) compile fine with 
your stable version.

regards
 frank
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[Asterisk-Users] Using a variable for EXTEN

Hi,

I tried set up a global var for an extension, like this

[globals]

IPPHONES=_3XX

[sip]

exten=${IPPHONES},1,Answer

What I would like to do is to make a dialplan without fixed extension
numbers to change the entire dialplan according to the customer
requests: what exten number do you want for your IP Phones ? let me change
a variable and we are set!

It seems that this is not supported, am I getting somethig wrong in
the syntax? There is another way to accomplish that ?

Tnx!

  

-- 
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Re: Grandstream Bugetone 101 mwi

2005-01-17 Thread Mark Elkins

On Sat, 2005-01-15 at 09:09 -0500, Doug Lytle wrote:
 Mike Dent wrote:

 Whilst on the subject of BT's, do the callers and called buttons function?
 they dont seem to do anything on mine?

 Yes, but the hand set needs to be off hook.

To add to Doug's reply...

---for people you have called---
1 - Pick up phone (or push 'speakerphone')
2 - Push 'called' - keep pushing it again and again - the displayed
number should change and the location where the time is usually
displayed will also change (increment)...
3 - When you get to the number you wish to call again - push 'send'

For people who have called you - exactly the same - except push the
'callers' button. The trick here is to make sure that the caller-id info
that the phone has saved (the people who have called you) somehow can be
sanely understood by your dial-plan logic..

I believe this works for the last 20 'called' and the last 20 'callers'.

Only flaw in the logic is that it would be nice to  push the
callers/called button - select the appropriate number and then when
pushing either 'send' or 'speakerphone' - activate the speakerphone and
dial the number...

-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [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] Using a variable for EXTEN

2005-01-17 Thread Dave Cotton

On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
 Hi,
 
 I tried set up a global var for an extension, like this
 
 [globals]
 
 IPPHONES=_3XX
 
 [sip]
 
 exten=${IPPHONES},1,Answer
 
 What I would like to do is to make a dialplan without fixed extension
 numbers to change the entire dialplan according to the customer
 requests: what exten number do you want for your IP Phones ? let me change
 a variable and we are set!
 
 It seems that this is not supported, am I getting somethig wrong in
 the syntax? There is another way to accomplish that ?

Works like that for me (tm).  If it's not working look carefully at all
your config files.


-- 
Dave Cotton [EMAIL PROTECTED]

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[Asterisk-Users] error compiling

2005-01-17 Thread Alberto Martnez

I got this error while compiling:

configure: error: termcap support not found

I don't know how to solve this problem...

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Re[2]: [Asterisk-Users] Using a variable for EXTEN

Hello Dave,

Monday, January 17, 2005, 12:50:13 PM, you wrote:

DC On Mon, 2005-01-17 at 12:30 +0100, Alessio Focardi wrote:
 Hi,
 
 I tried set up a global var for an extension, like this
 
 [globals]
 
 IPPHONES=_3XX
 
 [sip]
 
 exten=${IPPHONES},1,Answer
 
 What I would like to do is to make a dialplan without fixed extension
 numbers to change the entire dialplan according to the customer
 requests: what exten number do you want for your IP Phones ? let me change
 a variable and we are set!
 
 It seems that this is not supported, am I getting somethig wrong in
 the syntax? There is another way to accomplish that ?

DC Works like that for me (tm).  If it's not working look carefully at all
DC your config files.


I'm doing this using realtime, so really the dialplan is


[globals]

IPPHONES=_3XX

[sip]

switch=Realtime/sip


Then in the db I have the extentions using the variable name.

Someone ever tried this with realtime ?


-- 
Best regards,
 Alessiomailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] error compiling

2005-01-17 Thread Alberto Martnez

Sorry... I have forgotten to say I am compiling from sources
downloaded from the asterisk web page.

AM I got this error while compiling:

AM configure: error: termcap support not found

AM I don't know how to solve this problem...

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Re: [Asterisk-Users] TDM400 lost after reboot

2005-01-17 Thread Greg - Cirelle Enterprises

do a google search for
tdm400p hardware problems (fix)
This is a problem with the tdm card and driver
If you are using the older zaptel software the
file referenced in the doc is wcfxs.c if you
are using the cvs version the wcfxs file needs
to be replaced with wctdm.c also the line number
2127 is changes in the wctdm file so do a search
for pci_device_id or go to approx line 2130
HTH
Greg
At 09:37 AM 1/16/05, you wrote:
Hi
My card is working, but when I reboot the machine, most of the times
it is not working,
I get ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address 
(6)

To make it work again I have to shut down, remove the card, reboot so
kudzu will remove the config. shut down again, put the card back in,
reboot, now kudzu see it, I choose Ignore and then it's working
again (until the next reboot).
I'm on WBEL 3.0 and the card is not sharing is IRQ.
Is anybody else having this problem ?
When kudzu see it (as a Jens Schoenfeld Intel 537), what should I choose ?
Is there something I can do to prevent this from happening ?
Thanks
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Greg Cirino
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[Asterisk-Users] Manager Event Logging

2005-01-17 Thread Asterisk

Is there any way of logging all manager events to a file, similar to the 
entries in logger.conf.

I was actually hoping that there was such an entry in the logger.conf
ManagerEvent = root,rootevents
This would allow someone to interrogate all events for a given user (in 
this case root) from a file.

Is it possible ?
Julian
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Re: [Asterisk-Users] VOIP - INBOUND Call - best setup

2005-01-17 Thread Matthew Crocker

On Jan 16, 2005, at 8:45 PM, Joseph wrote:
What would be my best option to receive calls via VOIP.
I would like to use it as an alternative number when my main number is
busy.
The solution is not that easy as in order for customer to be a free 
call
DID=Direct Inward Dialing provider would need to be a local company, I
think.  Correct my anybody if I'm wrong.
I'm located in Alberta Canada so my chases are even smaller.
Not necessarily.  You could have your main number (PSTN) setup to call 
forward on busy to your VoIP number.  If the VoIP number is long 
distance you will pick up the LD charges on your main number.  Your 
customers would only need to know and call your main number.


I've another incoming fax line, so I guess I could set it somehow as an
alternative incoming line if my main line is busy.
--
#Joseph
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Re: [Asterisk-Users] chan_capi-0.3.5 error 127

2005-01-17 Thread Vincent Guidoux

Thank frank!

Now i have a un new prob

Went I make a call, the CAPI channel error

Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in
new stack Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request:
didn't find capi device with outgoing msn = 4202270. you should check
your config Jan 17 13:14:39 NOTICE[4146]: app_dial.c:746 dial_exec:
Unable to create channel of type 'CAPI'

salutations


On Mon, 17 Jan 2005 12:24:50 +0100, Frank Sautter [EMAIL PROTECTED] wrote:
 hi vincent,
 
 Vincent Guidoux schrieb:
  I have a problem for install chan_capi
  My pc: Suse 9.1, with asterisk current stable en cvs
  And patch the chan_capi
   chan_capi.c:1076: error: structure has no member named 'cid'
 as you are writing and apparent to the error message you are posting,
 you are using a stable 1.0.x version of asterisk.
 therefore you don't need to apply my patch, which is only for the
 HEAD-cvs version of asterisk.
 the version 0.3.5 from junghanns.net will (hopefuly) compile fine with
 your stable version.
 
 regards
   frank
 
 
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[Asterisk-Users] Can I start recording channel in the middle of conversation ?

2005-01-17 Thread Robert Rozman

Hi,

I'd kindly ask for simple example if this is possible ?

Is any key press encountered during conversation and action taken in
dialplan ?

Thanks,

regards,

Rob.

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[Asterisk-Users] Can I start recording channel in the middle of conversation ?

2005-01-17 Thread Robert Rozman

Hi,

I'd kindly ask for simple example if this is possible ?

Is any key press encountered during conversation and action taken in
dialplan ?

Thanks,

regards,

Rob.

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RE: [Asterisk-Users] ASTCC

2005-01-17 Thread Sebastian Atala



Here is the link 
http://www.voip-info.org/wiki-ASTCC

SA


-Mensaje original-
De: Bilal Ghayad [mailto:[EMAIL PROTECTED] 
Enviado el: Martes, 14 de Enero de 2003 18:21
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] ASTCC

Dear Sebastian;

Thanks a lot for your kindly advise to use ASTCC.

But can u advise me the link for ASTCC to download it and wether it is open
source (to download the source and work on it?

Regards
Bilal

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Re: [Asterisk-Users] No compatible codecs

2005-01-17 Thread Rene Kluwen

I am using the G729 stack from Intel with *.

But as far as I know the Grandstream can just connect with PCMU and * will
transcode the audio into G729, right?
Because I know that iaxcomm and SJPhone for sure do not support G729 but I
can connect with those clients.

Maybe I can try to completely disable the G729 stack in the Grandstream
*ponders* in case the problem is there.

Rene Kluwen
Chimit


- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 2:04 AM
Subject: Re: [Asterisk-Users] No compatible codecs


 I've heard problems with the Grandstream G729 and the new digium G729
 by MAC ID. Could be a compatibility issue with the implementations.
 Did you ever use the Grandstream against asterisk with the old
 Voiceage G729? I've heard that works just fine.

 -- William
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RE: [Asterisk-Users] pattern matching problem

2005-01-17 Thread Robert Jackson



 -Original Message-
 From: Joseph [mailto:[EMAIL PROTECTED] 
 Sent: Monday, January 17, 2005 1:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] pattern matching problem
 
 
 How do I solve the problem with between patterns:
 _1800
 _1NXX
 
 I would like all numbers 1800, 1877 etc to go through iaxtel 
 but all other numbers 1xxx via voipjet
 
In your default context (i.e. the one specified in sip.conf/iax.conf) 
include the iaxtel context before the outgoing-voipjet context.  The 
system should stop at the first match.

Good luck,

Robert Jackson
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RE: [Asterisk-Users] Can I start recording channel in the middle ofconversation ?

2005-01-17 Thread Robert Jackson

 -Original Message-
 From: Robert Rozman [mailto:[EMAIL PROTECTED] 
 Sent: Monday, January 17, 2005 7:45 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Can I start recording channel in 
 the middle ofconversation ?

 Hi,

 I'd kindly ask for simple example if this is possible ?

 Is any key press encountered during conversation and action 
 taken in dialplan ?

This is possible with CVS-HEAD within the last couple of weeks.  
The configuration is in features.conf, and I believe that *2 
activates the recording, but I could be wrong.  

Robert Jackson
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[Asterisk-Users] voicemail sound distorted - chan_capi, diva, cvs-head

2005-01-17 Thread Frank Sautter

hi,
i have a problem with distorted voicemail sound on our asterisk test 
machine.

i'm using cvs-head (2004-01-17) and chan_capi 0.3.5 (with my patches to 
make chan_capi compile with asterisk cvs-head) and a diva quad-bri isdn 
card.

other things work well with my setup (dial in, dial out, app_meetme) and 
sound recordings from sip channels.

the problem is with voicemail and app_record, where only a distorted 
sound can be heard in the recording if one shouts into the microphone of 
the telephone.

has anybody had the same problem or can confirm this issue?
regards
 frank
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Re: [Asterisk-Users] error compiling

2005-01-17 Thread Steven Critchfield

On Mon, 2005-01-17 at 12:56 +0100, Alberto Martínez wrote:
 I got this error while compiling:
 
 configure: error: termcap support not found

get termcap development package installed for your distribution.

-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] chan_capi outgoing msn

2005-01-17 Thread Frank Sautter

Vincent Guidoux schrieb:
Now i have a un new prob
Executing Dial(SIP/2500-0bbb, CAPI/@4202270:0796273153|30|r) in new
stack
Jan 17 13:14:39 NOTICE[4146]: chan_capi.c:1173 capi_request: didn't find
capi device with outgoing msn = 4202270. you should check your config
well the error message says it all. 'you should check your config'
apparently you haven't configured your MSNs in /etc/asterisk/capi.conf.
 8  snip
[interfaces]
msn=4202270
incomingmsn=*
 8  snap
regards frank
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Re: [Asterisk-Users] France has their (first?) SIP carrier with unlimited calls for 6eu/mo

2005-01-17 Thread Wilson Pickett

 /hax0r n00b mode on
 Which command and parameters do I need to use to get some legible (usable)
 output to do the packet sniffing? I tried ethereal but it only gives me
 loads of garbage?
 /hax0r n00b mode off  :)

Go to the Wengo forum, there is a thread in the technique section that
gives the entire process.
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RE: [Asterisk-Users] Operator Panels?

2005-01-17 Thread Matt Schulte

on 'make' 

chan_sccp.c: In function `load_module':
chan_sccp.c:653: warning: passing arg 4 of `ast_channel_register_ex'
from incompatible pointer type
Now compiling  sccp_actions.c   743 lines
Now compiling  sccp_channel.c   279 lines
sccp_channel.c: In function `sccp_channel_send_callinfo':
sccp_channel.c:48: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
sccp_channel.c:49: structure has no member named `callerid'
make: *** [.tmp/sccp_channel.o] Error 1

I would have posted in the sccp bugtracker but it looks like no one has
used it in months. This is from cvs compile. We tried using stable ast
(ie 1.0.2) but it broke our SIP (digest) authentication going to SER for
some reason.

Thanks, Matt

-Original Message-
From: Julien Goodwin [mailto:[EMAIL PROTECTED] 

What was your problem with chan_sccp? There's only one small issue I
know of in the code (already fixed, I just haven't committed it to CVS).

Although the biggest issue with using it would be that chan_sccp doesn't
yet have hint support (it's forthcoming once I get my new phone
delivered this week). 

Thanks,
Julien Goodwin 
chan_sccp developer
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Re: [Asterisk-Users] OT: SIP Aware Firewall with Asterisk

2005-01-17 Thread Jason Williams

On Mon, 10 Jan 2005 19:38:23 +, John Middleton
[EMAIL PROTECTED] wrote:
 Not an enterprise level system, but anyone used the www.intertex.se IX66?
 
Yes they work great no nat traversal issues,
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[Asterisk-Users] CAS voice signalling?

2005-01-17 Thread Daniel Nyström

According to CarrierAccess, the Adit 600 uses CAS for voice signalling. What is 
this?
This should not be a problem for Asterisk?
Does the Asterisk server need to reencode CAS into aLaw when going to Euro ISDN?

BR
Daniel Nyström
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Re: [Asterisk-Users] Softphone for Linux recommendation

2005-01-17 Thread Brian Johnson

I use iaxcomm (a little finicky to get working since it needs wxgtk files) and
linphone

Linphone is SIP and is a little trickier to get working when dealing with NAT
than iax (iaxcomm)

I chose these 2 because they seemed the easiest to get working.

They both work fine but are not as good sound quality as my hardware phones.



Howard Lowndes ([EMAIL PROTECTED]) wrote:

 Can anyone _recommend_ a downloadable OSS softphone that _works_ under
 Linux and is compatible with Asterisk.

 So far I have tried kphone and linphone and had problems with both, and
 I am still waiting to hear back from the X-Lite beta folks.

 when you want a system that just works, you choose Microsoft.
 --
 Flatter government, not fatter government;
 Get rid of the Australian states.


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Re: [Asterisk-Users] Re: Grandstream Bugetone 101 mwi

2005-01-17 Thread Mark Elkins

Curiosity got hold of me. I opened up my BT-10 (and it still works
afterwards..)

Under the keyboard (buttons) are four red LED's that appear to run in
parallel (they all flash at the same time when you put the power on).
These are used to light up the keyboard.

The Display LED (blue in my case) is flashed to indicate that a Message
is waiting

There appears to be no other LED's (or light sources) so no button will
ever (or can ever) flash...

In order to get the message button to work - programme it with the
extension number for your voice-mail.  On your BT-100's phone web page -
it looks something like.. 

Voice Mail UserID:[300]   (User ID/extension for 3rd party voice
mail system)  

So if I push the 'Message' button - I effectively dial '300' (ie the
same as picking up the handset and dialing '300').  In my
extensions.conf file - the appropriate line is... 

; 300 = Access Voicemail
; My 'Grandstreams' have a Message button - that I have programmed to
dial '300'
; They then pass over their CLID - so get to the correct mailbox
exten = 300,1,VoicemailMain(s${CALLERIDNUM})
exten = 300,2,Hangup

This will contact the Voicemail menu system - passing it the ID of the
phone that is calling it - the 's' is to skip the password
authentication..
Every BT-100 phone is set up in the same way - with the same '300' in
the Message Button field.

I also have the following set...  to **YES**

SUBSCRIBE for MWI:
Yes, send periodical SUBSCRIBE for Message Waiting Indication


So, with reasonably new firmware - the only button that does not seem to
have a function is 'Conference'. The 'Transfer' button is used for
attended (non-blind) transfers (see postings elsewhere).


On Fri, 2005-01-14 at 23:47 -0700, Paul Fielding wrote: 
  Hahawell the MWI is the blinking blue LCD.  The message button
  is reserved for future use  Hang in there.  There will soon to be some
  upgrades and rumor has it that the conferencing feature will soon be
  introduced so that conference button on the phone will soon be 
  working.
 
 The message button isn't reserved, it works fine, you simply need to 
 correctly configure it.   It's job is to dial the voicemail box when 
 pressed.   This works as designed.   It just doesn't blink.

  On Fri, 14 Jan 2005 10:25:46 -0500, Stephen R. Besch wrote
  Ronald Wiplinger wrote:
   I tried to use message waiting indicator, by Subscribe for MWI in the
   web menu of the phone.
  
   However, it does not light up / flash, even if a voice mail is waiting.
  
   Where is the switch to turn it to?

  I don't mean to be rude to everyone who responded to this question,
   but I think that everyone is answering the wrong question. The
  point is that the message waiting indicator doesn't light up, at all,
   ever. All that happens when messages are waiting is that the
  display blinks and the phone gives a stutter dialtone. That's it.
  There is no light under the button - there should be, but there
  isn't. The blinking phone designers should have put those stupid
  blinking red leds - that only flash on boot up - under the message
  button and flashed the display during boot up. But they didn't and
  we're stuck with it. Such is life.

-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [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] DIAX 0.9.9g more features and higher stability

Hi Dan, Steve, Michael, Bruno and others.

I will try to describe my VoIP environment below:

SERVER:
- FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
- iax.conf
[general]
bindport = 4569
bindaddr = 0.0.0.0
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
tos=lowdelay
jitterbuffer=no
dropcount=2
maxjitterbuffer=100
maxexccessbuffer=100
mailboxdetail=yes

[1001]
callerid=Ramal 1001 1001
context=from-internal
host=dynamic
mailbox=1001
notransfer=yes
port=4569
secret=
type=friend
username=1001

[1002]
callerid=Ramal 1002 1002
context=from-internal
host=dynamic
mailbox=1002
notransfer=yes
port=4569
secret=
type=friend
username=1002

CLIENT 1001:
- Windows XP
- DIAX 0.9.9g
- Firefly 1.9.6 Build 3944
- USB Phone NTP200E - Compatible with ATCOM USB Phone
- AMD 1.8Ghz with 256Mb

CLIENT 1002:
- Windows XP
- DIAX 0.9.9g
- Firefly 1.9.6 Build 3944
- USB Phone NTP200E - Compatible with ATCOM USB Phone
- AMD 1.66Ghz with 256Mb


ADDITIONAL INFORMATION
- All machines are in the same network(192.168.*.*) no firewall in the 
middle;
- With Firefly I have a VERY GOOD conversation, without any delay;
- With DIAX I have a one way delay of 10 sec. Only the person who recieve 
the call get the delay, the person who make the call listen without 
problems;
- Firefly in one side and DIAX in the other side, same delay problem;
- No problems with SIP;
- No problems(delay) with Linux clients runnig IaxComm 0.99pre11;
- Same problem with DIAX oldest DLL;
- Ping from clients to server: 0% packet loss and  1ms;
- No problems calling PSTN, Voicemail, etc, just between DIAX clients;

If you need something else, let me know!

Thanks for your help!

Denis Galvão.



Em Dom 16 Jan 2005 19:52, Steve Kann escreveu:
 On Jan 16, 2005, at 2:53 PM, Dan wrote:
  Hi Steve,
 
  - Original Message - From: Steve Kann [EMAIL PROTECTED]
 
  On Jan 14, 2005, at 2:03 PM, Dan wrote:
  Hi,
 
  \ Em Sex 14 Jan 2005 16:43, Dan escreveu:
   I dont have problems when calling PSTN extensions, and calling
   VoceMail, EchoTest, etc. The problem is related with the
 
  conversation
 
   between two DIAX Softphones.
 
  Between 2 DIAX phone and the delay is in one direction only??
 
  Yes. One direction only... Just who make the call get the delay.
 
  Then try
  jitterbuffer=no
  in iax.conf
  to see if it solves this issue.
 
  Dan et. al,
  I think this might be a problem with native transfers, and needing to
  reset the jitterbuffer history when this happens, or something like
  this..
  -SteveK
 
  But I have tried and I do don't have this problem here...
  What can I do to make this happen here?

 I don't know...

 Maybe if we could get a packet trace of the situation that causes the
 problem?

 Maybe try notransfer or whatever the iax.conf parameter is, and see if
 that changes things.  If it does, it points towards this being the
 problem.

 If the delay goes down after a couple of minutes after the transfer,
 this could be the problem.  If it doesn't, there's something else
 really wrong..

 (I'm assuming you're using the new JB code here..).  Also, if you're
 using the new JB code, you should implement the stuff to get the
 network stats, so we can see if calculated jitter is substantially
 higher..)

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RE: [Asterisk-Users] CAS voice signalling?

2005-01-17 Thread Jim Van Meggelen

[EMAIL PROTECTED] wrote:
 According to CarrierAccess, the Adit 600 uses CAS for voice
 signalling. What is this? This should not be a problem for
 Asterisk? Does the Asterisk server need to reencode CAS into aLaw
 when going to Euro ISDN? 

Try this:

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a
00800e2560.shtml

Cheers,

Jim.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.13 - Release Date: 16/01/2005
 

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Re: [Asterisk-Users] Gentoo CVS installation; was IAX1 vs. IAX2

2005-01-17 Thread Joseph

  I'm just using default installation whatever Gentoo is providing; this
  is their stable version.
  
 
 Joseph,
 
   While I also use Gentoo(as do many others), most will tell you NOT to 
 install * from portage.  You can save yourself trouble by getting 1.0.3 
 or CVS and ditch the builds from portage.
 
 --
 Kristian Kielhofner

How do you solve the dependency problem? 
Isn't the version in portage the same 1.0.3 as CVS?

I've tried to install ver.1.0.3 from portage before and there is one
package unstable still in unstable version that needed to be install
before 1.0.3

-- 
#Joseph
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Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread Andrew Kohlsmith

On January 17, 2005 01:47 am, John Sellens wrote:
 Just on the off chance that Canadian Asterisk users might be
 interested in a place to discuss topics specific to the great
 white north (sources, services, telcos, etc.), I created
 the asterisk-canada mailing list:

I know as a Canadian I'm not interested in a list Just for Canadians -- It's 
just fragmenting the help available for very little benefit.  I do, however, 
appreciate the thought.

-A.
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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread Gyrion, Larry M.

I'd be interested in a possible mailing list for a United States
Asterisk mailing list.  We are in the very beginning stages of building
a pilot system using Asterisk, but based on the information I've found
on the internet so far, it looks very promising to scale the system to
our needs. 
I'd be interested in know if anyone has successfully created a larger
system (at least 1000 to 1500 lines), and with a redundancy built in.  

Thank you,
Larry Gyrion
Telecommunications Administrator
Manchester College
604 East College Ave
North Manchester, IN  46962

-Original Message-
From: John Sellens [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 17, 2005 1:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Any interest in a Canadian Asterisk mailing
list?

Just on the off chance that Canadian Asterisk users might be
interested in a place to discuss topics specific to the great
white north (sources, services, telcos, etc.), I created
the asterisk-canada mailing list:
http://lists.syonex.com/mailman/listinfo/asterisk-canada
or
[EMAIL PROTECTED]

Cheers!

John

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[Asterisk-Users] REALTIME and VARIABLES

Hi,

I'm having some problem with realtime:

let's say I have a dialplan like this

[globals]

IPPHONES=_3XX

[sip]

exten=${IPPHONES},1,Answer
  
A call from ip phone 300 comes in, and it's been answered.

Then I switch the sip context to realtime, putting the exten in the
db and using the variable name for this as in the file version.

[globals]

IPPHONES=_3XX

[sip]

switch=Realtime/sip

Calling does not work anymore, extension is not found.

So it seems me that with realtime we cant'use variables as extensions
for an easyer manteniance of the dialplan.

Am I getting it all wrong ?

Tnx for any suggestion!



-- 
Best regards,
 Alessio  mailto:[EMAIL PROTECTED]

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[Asterisk-Users] Communication Between Phones... I can't test :(

2005-01-17 Thread Jeremy SALMON

Hi,

I want to create this system :


Desk1 SIP Phone adsladsl Desk2 SIP Phone
  |
  |
 adsl
 Desk3 asterisk Server



My question is : when Desk1 call Desk2 , server (desk3) will
authentificate phone but i want to known if Desk3 use bandwitch during
communication?

Thanks,

Jeremy



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[Asterisk-Users] ntp Server and Zultys 4X4

2005-01-17 Thread Ronald Hartmann

Good Day List,

I have my asterisk box setup to be an ntp server, and my zultys
4X4 phone  is able to get the time, however 
I must first select the TimeZone Offset and then it will use the
time setting from my server.

This is a hassle because every time the phone reboots the user
must answer this question and as you can imagine 
End users do not know what to do and since the phone is not
booted they can not call helpdesk..

Is there anyway to fix this.  Please excuse my ignorance if this
is an ntp server option I am unaware of.

ron


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Re: [Asterisk-Users] ntp Server and Zultys 4X4

2005-01-17 Thread Bruce Komito

For what it's worth, I'm working with Zultys trying to solve this exact
same problem. So far, they've told me to take an ethernet trace, because
they claim the DHCP option 42 isn't being sent, but I know this is not the
case, because the phone knows the time, just not the time zone. There is
a setting in the general section of the config file called timezone, which
defaults to -480 (minutes off of GMT), but this setting only seems to
control the value that you are prompted with when the phone boots.

If I get a solution, I'll let you know.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815

On Mon, 17 Jan 2005, Ronald Hartmann wrote:

Good Day List,

I have my asterisk box setup to be an ntp server, and my zultys
4X4 phone is able to get the time, however
I must first select the TimeZone Offset and then it will use the
time setting from my server.

This is a hassle because every time the phone reboots the user
must answer this question and as you can imagine
End users do not know what to do and since the phone is not
booted they can not call helpdesk..

Is there anyway to fix this. Please excuse my ignorance if this
is an ntp server option I am unaware of.

ron

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If you do not agree, please click on the link below to train the Analyzer.
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RE: [Asterisk-Users] ASTCC single stage + no access number + auth usingsip username and password

 I would like to have all SIP phones to work on prepaid basis
 and without need to dial any access number, instead I would
 like to use the phone as normal dialing only the destination
 number, for example 00464090510.

I use the AccountCode for authentication. This is how, for example:

exten = _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2})

 Once the call is finished I would like to have the balance
 shown in the display by sending a sip message to the phone(if
 possible otherwise not important).

This would require adding code to the AGI, if it's even possible.

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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Re: [Asterisk-Users] ASTCC single stage + no access number + auth usingsip username and password

2005-01-17 Thread Brian Wilkins

I believe you can specifiy header information. If you know what portion of 
packet deals with that information, you (in theory) would be able to do it. 

On Monday 17 January 2005 02:52 pm, Nabeel Jafferali wrote:
  I would like to have all SIP phones to work on prepaid basis
  and without need to dial any access number, instead I would
  like to use the phone as normal dialing only the destination
  number, for example 00464090510.

 I use the AccountCode for authentication. This is how, for example:

 exten = _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2})

  Once the call is finished I would like to have the balance
  shown in the display by sending a sip message to the phone(if
  possible otherwise not important).

 This would require adding code to the AGI, if it's even possible.

-- 
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]

Heritage Communications Corporation
  Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net

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Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread timebandit001

 I know as a Canadian I'm not interested in a list Just for Canadians -- It's
 just fragmenting the help available for very little benefit.  I do, however,
 appreciate the thought.
 
 -A.
I'm a Canadian also, and I second that
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[Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread R A

Hi all

I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.
I need to now the name os de file oraspecific category link where i can download it.

If you can send me the file is beter ;-)

Thanks inadvance
Regards

Wert
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Yahoo! Mail - Find what you need with new enhanced search. Learn more.___
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[Asterisk-Users] RE: [Asterisk-biz] Guatemala DID's?

2005-01-17 Thread Tenorio, Leandro


In the next couple of weeks we will be starting the beta phase of our
Guatemala POP. If you could wait, welcome.

LTenorio

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Phil Astin
Sent: Sunday, January 16, 2005 6:23 PM
To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com
Subject: [Asterisk-biz] Guatemala DID's?

I'm looking for a company that offers Guatemala DID's. I saw that Lingo
does, but Lingo isn't easily compatible w/ Asterisk, so they're a last
resort.
Thanks in advanced, Phil Astin.
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Re: [Asterisk-Users] No compatible codecs (solved)

2005-01-17 Thread Rene Kluwen

I can now place calls from Grandstream (via Asterisk) to mutualphone.

I did this by disabling the G729 (and G723) codecs in the Grandstream, so
that * takes care of any recoding.
What it looks like is that the G729 stack of the BT101 is not compatible
with the one that mutualphone is using.

Either way, for me this solution works.

Rene Kluwen
Chimit

- Original Message -
From: Rene Kluwen [EMAIL PROTECTED]
To: William Suffill [EMAIL PROTECTED]; Asterisk Users Mailing
List - Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 1:47 PM
Subject: Re: [Asterisk-Users] No compatible codecs


 I am using the G729 stack from Intel with *.

 But as far as I know the Grandstream can just connect with PCMU and * will
 transcode the audio into G729, right?
 Because I know that iaxcomm and SJPhone for sure do not support G729 but I
 can connect with those clients.

 Maybe I can try to completely disable the G729 stack in the Grandstream
 *ponders* in case the problem is there.

 Rene Kluwen
 Chimit


 - Original Message -
 From: William Suffill [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Monday, January 17, 2005 2:04 AM
 Subject: Re: [Asterisk-Users] No compatible codecs


  I've heard problems with the Grandstream G729 and the new digium G729
  by MAC ID. Could be a compatibility issue with the implementations.
  Did you ever use the Grandstream against asterisk with the old
  Voiceage G729? I've heard that works just fine.
 
  -- William
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RE: [Asterisk-Users] SS7 and Asterisk solution

2005-01-17 Thread Michael Baird

Same here, interested in the details of a SS7/Asterisk solution.

Regards
MIKE
 Steve,
 
 I also would be very interested in getting those details. We would very
 much like to move forward with SS7, please feel free to contact me off
 list.
 
 Cheers,
 
 Ben Merrills
 Griffin Internet
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Felix
 Skwarczynski
 Sent: 14 January 2005 09:23
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SS7 and Asterisk solution
 
 Hi Steve,
 
 I also want the commercial details, so if you can send them to me or put
 
 me in touch with somebody who can it would be very helpfull.
 
 Thank you in advance,
 Felix Skwarczynski
 
 Steve Underwood wrote:
 
  Hi Bartosz,
 
  We have a commercial SS7 for Asterisk that is running at a few test 
  sites, and which we are just about ready to supply to a broader range 
  of customers. This actually links into Asterisk, so we need to use a 
  commercially licenced copy of Asterisk. If this sounds interesting to 
  you, I can put you in touch with someone who will give you the 
  commercial details.
 
  Regards,
  Steve
 
  Bartosz Jozwiak wrote:
 
  Hello,
 
  We are looking for commercial solution SS7 with Asterisk.
  It does not need to be build-in with Asterisk.
  Could anybody suggest something?
 
  Thank you in advance.
  Bart
 
 
 
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RE: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

 I was looking for the SIP IOS of the Cisco IP Phone but i
 can´t find it in the cisco web page.

What is IOS? Am I the only one who uses Cisco phones and doesn't know that 
acronym?

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

 I know as a Canadian I'm not interested in a list Just for
 Canadians -- It's just fragmenting the help available for very
 little benefit. I do, however, appreciate the thought.

I disagree. I have joined the new list and feel that as long as it is
focused on discussions like:

- DIDs in Canada
- VoIP taxes/regulation in Canada
- * compatible hardware vendors in Canada

then it should be a welcome addition to the * community. However, any
general question should be directed to the -users list.

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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Re: [Asterisk-Users] France has their (first?) SIP carrier with unlimited calls for 6eu/mo

2005-01-17 Thread Remco Barende

On Mon, 17 Jan 2005, Wilson Pickett wrote:
/hax0r n00b mode on
Which command and parameters do I need to use to get some legible (usable)
output to do the packet sniffing? I tried ethereal but it only gives me
loads of garbage?
/hax0r n00b mode off  :)
Go to the Wengo forum, there is a thread in the technique section that
gives the entire process.
One more question, if I try to dial the French telephone number I received 
I hear the tone that the number is not in use.

Is there a mistake in my asterisk config or is the French number only 
accessable from France?


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Re: [Asterisk-Users] REALTIME and VARIABLES

2005-01-17 Thread Matthew Boehm

 So it seems me that with realtime we cant'use variables as extensions
 for an easyer manteniance of the dialplan.

Doesn't RealTime itself make for easier maintenance of extensions since
its database driven?

-Matthew

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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

Two more information:

1. I've played with all suported codecs, same problems for all of them.

2. After aprox. 1 minute of conversation the delay problem doesn't occur, or 
better, it is very less(some miliseconds) than the begining(10 seconds) of 
a call.

Any ideas!?

Denis.


Em Seg 17 Jan 2005 11:51, Denis Galvão - iSolve escreveu:
 Hi Dan, Steve, Michael, Bruno and others.

 I will try to describe my VoIP environment below:

 SERVER:
 - FC1 with Asterisk CVS-v1-0-11/04/04-23:47:17
 - iax.conf
 [general]
 bindport = 4569
 bindaddr = 0.0.0.0
 delayreject=yes
 disallow=all
 allow=ulaw
 allow=alaw
 allow=gsm
 tos=lowdelay
 jitterbuffer=no
 dropcount=2
 maxjitterbuffer=100
 maxexccessbuffer=100
 mailboxdetail=yes

 [1001]
 callerid=Ramal 1001 1001
 context=from-internal
 host=dynamic
 mailbox=1001
 notransfer=yes
 port=4569
 secret=
 type=friend
 username=1001

 [1002]
 callerid=Ramal 1002 1002
 context=from-internal
 host=dynamic
 mailbox=1002
 notransfer=yes
 port=4569
 secret=
 type=friend
 username=1002

 CLIENT 1001:
 - Windows XP
 - DIAX 0.9.9g
 - Firefly 1.9.6 Build 3944
 - USB Phone NTP200E - Compatible with ATCOM USB Phone
 - AMD 1.8Ghz with 256Mb

 CLIENT 1002:
 - Windows XP
 - DIAX 0.9.9g
 - Firefly 1.9.6 Build 3944
 - USB Phone NTP200E - Compatible with ATCOM USB Phone
 - AMD 1.66Ghz with 256Mb


 ADDITIONAL INFORMATION
 - All machines are in the same network(192.168.*.*) no firewall in the
 middle;
 - With Firefly I have a VERY GOOD conversation, without any delay;
 - With DIAX I have a one way delay of 10 sec. Only the person who recieve
 the call get the delay, the person who make the call listen without
 problems;
 - Firefly in one side and DIAX in the other side, same delay problem;
 - No problems with SIP;
 - No problems(delay) with Linux clients runnig IaxComm 0.99pre11;
 - Same problem with DIAX oldest DLL;
 - Ping from clients to server: 0% packet loss and  1ms;
 - No problems calling PSTN, Voicemail, etc, just between DIAX clients;

 If you need something else, let me know!

 Thanks for your help!

 Denis Galvão.

 Em Dom 16 Jan 2005 19:52, Steve Kann escreveu:
  On Jan 16, 2005, at 2:53 PM, Dan wrote:
   Hi Steve,
  
   - Original Message - From: Steve Kann [EMAIL PROTECTED]
  
   On Jan 14, 2005, at 2:03 PM, Dan wrote:
   Hi,
  
   \ Em Sex 14 Jan 2005 16:43, Dan escreveu:
I dont have problems when calling PSTN extensions, and calling
VoceMail, EchoTest, etc. The problem is related with the
  
   conversation
  
between two DIAX Softphones.
  
   Between 2 DIAX phone and the delay is in one direction only??
  
   Yes. One direction only... Just who make the call get the delay.
  
   Then try
   jitterbuffer=no
   in iax.conf
   to see if it solves this issue.
  
   Dan et. al,
   I think this might be a problem with native transfers, and needing
   to reset the jitterbuffer history when this happens, or something
   like this..
   -SteveK
  
   But I have tried and I do don't have this problem here...
   What can I do to make this happen here?
 
  I don't know...
 
  Maybe if we could get a packet trace of the situation that causes the
  problem?
 
  Maybe try notransfer or whatever the iax.conf parameter is, and see if
  that changes things.  If it does, it points towards this being the
  problem.
 
  If the delay goes down after a couple of minutes after the transfer,
  this could be the problem.  If it doesn't, there's something else
  really wrong..
 
  (I'm assuming you're using the new JB code here..).  Also, if you're
  using the new JB code, you should implement the stuff to get the
  network stats, so we can see if calculated jitter is substantially
  higher..)
 
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-- 
D e n i s   G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu, 526 1206B
CEP: 80530-000 - Curitiba - PR
+55 41 252-2977
http://www.isolve.com.br



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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

2005-01-17 Thread Dan

Hi Denis,
From: Denis Galvão - iSolve [EMAIL PROTECTED]
...
- Same problem with DIAX oldest DLL;
It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0
Please try an older version of DIAX, like 0.9.8c.
You can still download it from:
http://www.laser.com/dante/diax/diax098c.zip
or even older:
http://www.laser.com/dante/diax/diax097a.zip
http://www.laser.com/dante/diax/diax096d.zip
http://www.laser.com/dante/diax/diax095.zip
and see if the problem persist.
If not, then it must be something in the new library and we will dig 
further.

Thank you and best regards,
Dan
P.S. Pls tell me the version working without delay... 

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RE: [Asterisk-Users] ntp Server and Zultys 4X4

2005-01-17 Thread Ronald Hartmann

I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on
this and I think the issue may be related to the setting of the Time
Offset 

3.4. Time Offset

   The time offset field specifies the offset of the client's subnet in
   seconds from Coordinated Universal Time (UTC).  The offset is
   expressed as a two's complement 32-bit integer.  A positive offset
   indicates a location east of the zero meridian and a negative offset
   indicates a location west of the zero meridian.

   The code for the time offset option is 2, and its length is 4 octets.

Code   LenTime Offset
   +-+-+-+-+-+-+
   |  2  |  4  |  n1 |  n2 |  n3 |  n4 |
   +-+-+-+-+-+-+

Once I have time to play with this I will check it out.. any
feedback is appreciated.

Ron

-Original Message-
From: Bruce Komito [mailto:[EMAIL PROTECTED] 
Sent: Monday, January 17, 2005 9:38 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] ntp Server and Zultys 4X4

For what it's worth, I'm working with Zultys trying to solve this exact
same problem.  So far, they've told me to take an ethernet trace,
because
they claim the DHCP option 42 isn't being sent, but I know this is not
the
case, because the phone knows the time, just not the time zone.  There
is
a setting in the general section of the config file called timezone,
which
defaults to -480 (minutes off of GMT), but this setting only seems to
control the value that you are prompted with when the phone boots.

If I get a solution, I'll let you know.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815


On Mon, 17 Jan 2005, Ronald Hartmann wrote:

 Good Day List,

   I have my asterisk box setup to be an ntp server, and my zultys
 4X4 phone  is able to get the time, however
   I must first select the TimeZone Offset and then it will use the
 time setting from my server.

   This is a hassle because every time the phone reboots the user
 must answer this question and as you can imagine
   End users do not know what to do and since the phone is not
 booted they can not call helpdesk..

   Is there anyway to fix this.  Please excuse my ignorance if this
 is an ntp server option I am unaware of.

 ron


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Re: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread Andrew Kohlsmith

On January 17, 2005 10:26 am, Nabeel Jafferali wrote:
 I disagree. I have joined the new list and feel that as long as it is
 focused on discussions like:

 - DIDs in Canada

That's a -biz question

 - VoIP taxes/regulation in Canada

While not specifically -biz, all that can be said on that at this point is 
that it's a gray area and the discussions relating to that would be contained 
in fewer than a dozen threads anyway.

 - * compatible hardware vendors in Canada

-biz again...

I dunno, those three items and all the discussion that could possibly entail 
them are hardly worth putting a completely separate mailing list together 
for.

 then it should be a welcome addition to the * community. However, any
 general question should be directed to the -users list.

Agreed -- I dunno until the Canuck discussions start overshadowing the -users 
traffic I think it's a poor idea to try and separate it out.  Just my 
CAD$0.02 though.

-A.
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RE: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread R A

the image file to get it working with asterisk

sorry for the acronym

wertNabeel Jafferali [EMAIL PROTECTED] wrote:
 I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.What is IOS? Am I the only one who uses Cisco phones and doesn't know that acronym?-- Nabeel JafferaliTel: +1 (416) 628-9342 Toronto+1 (646) 225-7426 New YorkFWD: 46990Email/MSN: nabeeljafferali.net___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

2005-01-17 Thread Steve Kann

Denis Galvão - iSolve wrote:
Two more information:
1. I've played with all suported codecs, same problems for all of them.
2. After aprox. 1 minute of conversation the delay problem doesn't occur, or 
better, it is very less(some miliseconds) than the begining(10 seconds) of 
a call.

Any ideas!?
 

Yes, it sounds like there's a discontinuity in the timestamps when you 
set up your call, but it seems Dan can't reproduce this.

The fix is probably:
a) The jitterbuffer needs to be reset after the transfer, or
b) The timestamps sent need to be reset after the transfer.
c) Some changes to the jitterbuffer to automatically reset when it sees 
this kind of discontinuity.

(c can probably be combined with a and/or b).
I forget if you tried setting notransfer=yes on asterisk to see what 
that does?

What would really help, though, is a packet trace of the call.   The 
best way to get this is to use either ethereal or tcpdump.  (there is an 
ethereal for windows). 

If you use ethereal for Windows, have it capture all udp, make the call, 
and have it stay up for about 30 seconds, and save the file.   You can 
then send that file to me, and I'll be able to see what's going on a lot 
better than guessing here..

-SteveK
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Re: [Asterisk-Users] CAS voice signalling?

2005-01-17 Thread Daniel Nyström

Oh, thanks!
How do I know which codec is used on Adit 600?
Does the server need to reencode it at all, or is the codec the same on Euro 
ISDN?
If it has to reencode everything, it really seems to be CPU critical when using 
30 FSX lines into 30 Euro ISDN lines.

Btw, when using Adit for connecting 30 handsets. Is it FXS or FXO modules I 
need? As I've seen, there is alot of misunderstanding in that particulary case.

BR
Daniel Nyström

- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, January 17, 2005 2:57 PM
Subject: Re: [Asterisk-Users] CAS voice signalling?


 On Mon, 2005-01-17 at 14:33 +0100, Daniel Nyström wrote:
  According to CarrierAccess, the Adit 600 uses CAS for voice signalling. 
  What is this?
  This should not be a problem for Asterisk?
  Does the Asterisk server need to reencode CAS into aLaw when going to Euro 
  ISDN?
 
 CAS is Channel Associated Signaling. It isn't dependent on alaw or ulaw
 it has to do with where the signaling bits are. 
 
 Adit 600's are well known to work with asterisk. 
 -- 
 Steven Critchfield [EMAIL PROTECTED]
 
 

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RE: [Asterisk-Users] NuFone help

2005-01-17 Thread Mark Halverson

I too had the same problem - it fixed itself the other day.

Of course, it was five days after reporting the problem with no response
from NuFone...additionally, if I attempted to call and # in the 707 area
code the call would not go through.

The other problem that I find with NuFone is the CLEC that they are using
does NOT recognize the new area code splits and their switch is NOT
programmed properly.  I have a nationwide 800# and noticed that when I
SetCallerID there are EIGHT area codes that I cannot call from. I am using
VALID ANIs because I can pickup the 'landline' phone with the same ANI and
call the number just fine.  What is strange is the there are other
nationwide 800# that I can call just fine and others that I can't.

Anyone else having that problem?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jake Franklin
Sent: Saturday, January 15, 2005 3:18 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NuFone help

Hello,

I've signed up for a NuFone account, and added the following 
instructions to my config files per NufFones directinos:

iax.conf
[NuFone]
type=peer
host=switch-1.nufone.net
secret=password

extensions.conf
(under the [default] context)
exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}

I then get this message in the CLI:

 -- Executing Dial(SIP/jake-fe5d, 
IAX2/user:[EMAIL PROTECTED]/1303555) in new stack
 -- Called user:[EMAIL PROTECTED]/1303555
 -- Call accepted by 66.225.202.72 (format gsm)
 -- Format for call is gsm
 -- Hungup 'IAX2/NuFone/1'
   == No one is available to answer at this time

I have, of course, changed the username/passwd and phone # for security 
reasons in this e-mail.

Any help would be greatly appreciated!

Jake
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Re: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread Shaun Ewing

On Mon, 17 Jan 2005 07:11:20 -0800 (PST), R A [EMAIL PROTECTED] wrote:
 Hi all
  
 I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in
 the cisco web page.
 I need to now the  name os de file or a specific category  link where i can
 download it.
  
 If you can send me the file is beter   ;-)
  
 Thanks in advance
 Regards
  
 Wert  

You're out of luck.

The 7902G only supports SCCP.

-Shaun
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Re: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread Shaun Ewing

On Mon, 17 Jan 2005 10:23:57 -0500, Nabeel Jafferali
[EMAIL PROTECTED] wrote:
  I was looking for the SIP IOS of the Cisco IP Phone but i
  can´t find it in the cisco web page.
 
 What is IOS? Am I the only one who uses Cisco phones and doesn't know that 
 acronym?

Internetworking Operating System.

It's what Cisco calls the operating system that runs on their routers, etc.

-Shaun
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Re: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread Christopher L. Wade

Nabeel Jafferali wrote:
I was looking for the SIP IOS of the Cisco IP Phone but i
can´t find it in the cisco web page.

What is IOS? Am I the only one who uses Cisco phones and doesn't know that 
acronym?
Internetwork Operating System (I. O. S. or IOS).  If I remember 
correctly, the phones don't run IOS, what they do run is very similar, 
but not quite IOS.  Kind of like their Wireless AP's use two different 
firmware's, VxWorks (sp?) and IOS.

Anyway, to the OP, I don't think the 7902G can run SIP.  Only the 
7940/7960 phones have the SIP option, again IIRC.

-Chris
~ Everyone's brain started out as a single cell. ~
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RE: [Asterisk-Users] DIAX 0.9.9g more features and higher stabili ty

2005-01-17 Thread Whisker, Peter

I have had the same problem when calling across Asterisk from Diax to a SIP
phone. If Asterisk Answers the call before the Dial to the SIP phone
there is no delay. Otherwise there is a 10-20 second delay in the Voice
path!

Peter 

-Original Message-
From: Dan [mailto:[EMAIL PROTECTED]
Sent: 14 January 2005 15:57
To: Denis Galvão - iSolve; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] DIAX 0.9.9g more features and higher
stability


Hi Denis,

 Doesn't have effect, the problem continue.
The strange thing is that the delay is aprox 10-20 seconds!!! Too high!!
Is there another thing to do!?

I really want to use DIAX because it supprts my USB Phone and IAX protocol.

I don't think your problem is DIAX related.
Can you provide more info about your environment?

In the Control Panel Sound Configuration have you selected MIC as only input

for the used soundcard and wave out as the output?

Best regards,
Dan


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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

 Just my CAD$0.02 though.

C'mon, at least throw in a loonie :P

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

 the image file to get it working with asterisk

You mean the SIP image? Look at the Wiki, there's info on what service
contract you have to buy from Cisco to get access to it. I believe
technically, although the service contract gives you access to the
image, you need a specific SIP image license to actually use it.

AND STOP USING HTML! :P

-- 
Nabeel Jafferali
Tel: +1 (416) 628-9342  Toronto
 +1 (646) 225-7426  New York
FWD: 46990
Email/MSN: nabeelatjafferali.net
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RE: [Asterisk-Users] SIP IOS for cisco 7902G IP Phone

2005-01-17 Thread R A

you can call it firmware upgrade for SIP

i just want to register this phone with asterisk, 
I apreciate any other idea to do it.

thanks in advance 
wert
Nabeel Jafferali [EMAIL PROTECTED] wrote:
 I was looking for the SIP IOS of the Cisco IP Phone but i can´t find it in the cisco web page.What is IOS? Am I the only one who uses Cisco phones and doesn't know that acronym?-- Nabeel JafferaliTel: +1 (416) 628-9342 Toronto+1 (646) 225-7426 New YorkFWD: 46990Email/MSN: nabeeljafferali.net___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

Em Seg 17 Jan 2005 13:51, Steve Kann escreveu:
 Yes, it sounds like there's a discontinuity in the timestamps when you
 set up your call, but it seems Dan can't reproduce this.

 The fix is probably:

 a) The jitterbuffer needs to be reset after the transfer, or
 b) The timestamps sent need to be reset after the transfer.
 c) Some changes to the jitterbuffer to automatically reset when it sees
 this kind of discontinuity.

 (c can probably be combined with a and/or b).

 I forget if you tried setting notransfer=yes on asterisk to see what
 that does?

Yes, Im using notransfer=yes, like my iax extension:

[1001]
callerid=Ramal 1001 1001
context=from-internal
host=dynamic
mailbox=1001
notransfer=yes
port=4569
secret=1001
type=friend
username=1001


 What would really help, though, is a packet trace of the call.   The
 best way to get this is to use either ethereal or tcpdump.  (there is an
 ethereal for windows).

 If you use ethereal for Windows, have it capture all udp, make the call,
 and have it stay up for about 30 seconds, and save the file.   You can
 then send that file to me, and I'll be able to see what's going on a lot
 better than guessing here..

Ok, I will do it.

Thanks Steve.

Denis.
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Re: [Asterisk-Users] DIAX 0.9.9g more features and higher stability

Em Seg 17 Jan 2005 13:43, Dan escreveu:
 Hi Denis,

 From: Denis Galvão - iSolve [EMAIL PROTECTED]
 ...
  - Same problem with DIAX oldest DLL;

 It is not an old DLL, but the same DLL build with NEW JITERBUFFER 0

 Please try an older version of DIAX, like 0.9.8c.
 You can still download it from:

 http://www.laser.com/dante/diax/diax098c.zip
 or even older:
 http://www.laser.com/dante/diax/diax097a.zip
 http://www.laser.com/dante/diax/diax096d.zip
 http://www.laser.com/dante/diax/diax095.zip

 and see if the problem persist.

 If not, then it must be something in the new library and we will dig
 further.

 Thank you and best regards,
 Dan
 P.S. Pls tell me the version working without delay...

Ok, Dan, I will try it out, and I'll inform you the results.

Denis.
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RE: [Asterisk-Users] DIAX

2005-01-17 Thread Whisker, Peter

GSM Codec is 13k plus overhead. That may work?

Peter

-Original Message-
From: Bilal Ghayad [mailto:[EMAIL PROTECTED]
Sent: 15 January 2005 07:07
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] DIAX

Dear Dan;

Thanks alot for your kindly reply.

Well, what u advise us to use if the bandwidth is about 22kbps (dial up
connection in very old countries)?

Another thing: u have idea if it is working on Microsoft Windows OS? As most
of clients here are using Microsoft and not linux.

Regards
Bilal

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Re[2]: [Asterisk-Users] REALTIME and VARIABLES

Hello Matthew,

Monday, January 17, 2005, 4:34:16 PM, you wrote:

 So it seems me that with realtime we cant'use variables as extensions
 for an easyer manteniance of the dialplan.

MB Doesn't RealTime itself make for easier maintenance of extensions since
MB its database driven?

So this is not a bug, it's a feature! :)

Seriously, anyone verified my problem and it's willing to share a
solution if there is any ?



-- 
Best regards,
 Alessiomailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Can't initiate a call with X-Lite.

2005-01-17 Thread Mario . Spoljar






 No. You should work on configuring xlite to register with asterisk.
 In the xlite Sip Proxy menu, you will need a User Name, Password,
 Sip Proxy, and Domain/Realm defined to match entries in your
 sip.conf definitions.


to which entry have to corespond Domain/Realm parameter in X-lite


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[Asterisk-Users] Is it possible to ID payphone calls?

2005-01-17 Thread Jess Coburn

Hello I have a 800 DID setup to dial into my Asterisk server and I'm
wondering if it's possible to ID when it's a payphone or not?  I
suspect it's not since I'm getting calls from someone else's SIP or
IAX box.

If I had a digium card installed and connected to a couple lines would
I be able to get this information and parse it?

Thanks,
Jess
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[Asterisk-Users] simple over view of the process

2005-01-17 Thread lonnie

Hello All,

Please forgive the lack of understanding as of yet but I have been trying
to follow the mailing list messages over the last few days and would like
to know if someone could wither point me into the right direction or
possibly give me a brief overview of the complete process.

Basically, I see that the Asterisk PBX systems can run on linux and seems
to offer the engine base that is needed for the SIP clients to connect.

Additionally, it seems that the various hardware (of which I have no idea)
if installed into the server will allow the SIP clients to communicate
with analog lines.

What inexpensive hardware is need to set up a basic system?

Thanks,
-Lonnie

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RE: [Asterisk-Users] ntp Server and Zultys 4X4

2005-01-17 Thread Bruce Komito

That was the hint I needed. Try adding this to your dhcp.conf:

option time-offset -480

(-480 is for PST, -420 is mountain, etc.)

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815

On Mon, 17 Jan 2005, Ronald Hartmann wrote:

I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on
this and I think the issue may be related to the setting of the Time
Offset

3.4. Time Offset

The time offset field specifies the offset of the client's subnet in
seconds from Coordinated Universal Time (UTC). The offset is
expressed as a two's complement 32-bit integer. A positive offset
indicates a location east of the zero meridian and a negative offset
indicates a location west of the zero meridian.

The code for the time offset option is 2, and its length is 4 octets.

Code LenTime Offset
+-+-+-+-+-+-+
| 2 | 4 | n1 | n2 | n3 | n4 |
+-+-+-+-+-+-+

Once I have time to play with this I will check it out.. any
feedback is appreciated.

Ron

-Original Message-
From: Bruce Komito [mailto:[EMAIL PROTECTED]
Sent: Monday, January 17, 2005 9:38 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] ntp Server and Zultys 4X4

For what it's worth, I'm working with Zultys trying to solve this exact
same problem. So far, they've told me to take an ethernet trace,
because
they claim the DHCP option 42 isn't being sent, but I know this is not
the
case, because the phone knows the time, just not the time zone. There
is
a setting in the general section of the config file called timezone,
which
defaults to -480 (minutes off of GMT), but this setting only seems to
control the value that you are prompted with when the phone boots.

If I get a solution, I'll let you know.

Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815

On Mon, 17 Jan 2005, Ronald Hartmann wrote:

Good Day List,

Is there anyway to fix this. Please excuse my ignorance if this
is an ntp server option I am unaware of.

ron

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RE: [Asterisk-Users] Any interest in a Canadian Asterisk mailing list?

2005-01-17 Thread Jim Van Meggelen

[EMAIL PROTECTED] wrote:
 On January 17, 2005 01:47 am, John Sellens wrote:
 Just on the off chance that Canadian Asterisk users might be
 interested in a place to discuss topics specific to the great white
 north (sources, services, telcos, etc.), I created the
 asterisk-canada mailing list:
 
 I know as a Canadian I'm not interested in a list Just for
 Canadians -- It's just fragmenting the help available for very
 little benefit. I do, however, appreciate the thought.

I don't think the idea is to be just for Canadians, but more as a
forum for topics that relate to Asterisk in the Canadian environment. 

A very relevant example is the CRTCs deliberations on VoIP, which may
have huge repercussions to Canadian Asterisk users, but is hardly
relevant to the international version of the Asterisk-Users list. Bell
and TELUS bashing might also be popular topics :-)

I do agree that any subject that is not specific to the Canadian
experience should remain in the international list. We are an
international community; therein lies our power.

Anyhow, I signed up, and am planning to start a thread about the CRTC
VoIP deliberations (and the generous act performed there by Jeff
Pulver), something I wouldn't feel was appropriate on Asterisk-Users.
Time will tell how many topics there are to discuss.

The way I see it, a Canadian mailing list will be no different than our
country itself: visitors will always be welcome.


--
Jim Van Meggelen
[EMAIL PROTECTED]
 

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Re: [Asterisk-Users] simple over view of the process