Re: [Asterisk-Users] (no subject)

[Jordan Peterson]

I don't know what that is, so probably not.  Is that a conference type
board?  Is there a way to make conferencing work or to assign an
extension to a h323 connection?

Thanks

On Mon, 2003-06-23 at 23:42, Jeremy McNamara wrote:
> Do you have a Zaptel device in this machine?
> 
> Jeremy McNamara
> 
> 
> 
> 
> 
> 
> Jordan Peterson wrote:
> 
> >Is this me or what?
> >
> >-- Playing 'demo-congrats'
> >-- Executing MeetMe("H323:996", "") in new stack
> >-- Playing 'conf-getconfno'
> >  == Parsing '/etc/asterisk/meetme.conf': Found
> >WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open
> >pseudo channel
> >-- Playing 'conf-invalid'
> >-- Playing 'conf-getconfno'
> >-- Playing 'conf-getconfno'
> >
> >
> >Thanks
> >
> >  
> >
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Jordan


In a world without windows, who needs gates?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] asterisk and passwords

[Rainer Jochem]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi.

Perhaps I just didn't found it, but there are two things I'm currently missing
in asterisk:

- -   The user passwords for SIP are written *plain ascii* in the   
sip.conf... is there any possibility to store them encryptet?
- -   How can a user change his(her) SIP-Password? Is there any key in
the voicemail/agent-system which allows changing the SIP-Password?
The only thing I've found there is the possibility of changing the
voicemail password. Or exists somewhrer an web-based form (like the
SIPExpressRouter has) to do so?

Thanks for your help
 Rainer
- -- 
http://graphics.cs.uni-sb.de/~rainer/
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.0.6 (GNU/Linux)
Comment: For info see http://www.gnupg.org

iD8DBQE+9/fKH/5+6U9F1qYRAnlLAJwN2sjtyXLwR1LzCqBu7Xg+EltuzQCfaDyU
y39p7YeM5X8yhNtt7D+7veo=
=Um1g
-END PGP SIGNATURE-

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] databases for billing

[Ing. Angel Gomez Garcia]

hostname=localhost
dbname=asteriskcdrdb
password=
user=asteriskcdruser


carlos del mayor wrote:

  I'm only asking for some examples of cdr_mysql.conf, nobody has
done anything with cdr and mysql? If you think is better another DB,,,
tell me, please!
  thanks in advance
  carlos
  
  carlos del mayor <[EMAIL PROTECTED]> wrote:
  
can you be more explicit, please? or give me some examples?
please, i'm little lost!
thanks a lot
carlos

Martin Pycko <[EMAIL PROTECTED]> wrote:
cdr_mysql.conf
  
On Fri, 20 Jun 2003, carlos del mayor wrote:
  
> hi
> I want to do a database to save the cdr with a billing finality.
I've created the database in mysql (thanks for the table and all that!)
but I'm not sure of how to 'connect' asterisk to that database in order
to save there the cdr. Is the cdr_mysql.conf what I have to config? Or
must I do a script, with the 'database' AGI commands?
> Any help would be so apreciated!
> Thanks a lot
> carlos
>
>
> -
> Yahoo! Sorteos
> Juega a la Lotería Primitiva sin salir de casa
  
  
  
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 Ya!
hoo! Sorteos
Juega
a la Lotería Primitiva sin salir de casa
  Yahoo!
Sorteos
  Juega
a la Lotería Primitiva sin salir de casa 

Re: [Asterisk-Users] (no subject)

[Ing. Angel Gomez Garcia]

You must have a Zaptel device installed in your computer or load  
ztdummy module to get conferencing to work...

Jordan Peterson wrote:

I don't know what that is, so probably not.  Is that a conference type
board?  Is there a way to make conferencing work or to assign an
extension to a h323 connection?
Thanks

On Mon, 2003-06-23 at 23:42, Jeremy McNamara wrote:
 

Do you have a Zaptel device in this machine?

Jeremy McNamara





Jordan Peterson wrote:

   

Is this me or what?

  -- Playing 'demo-congrats'
  -- Executing MeetMe("H323:996", "") in new stack
  -- Playing 'conf-getconfno'
== Parsing '/etc/asterisk/meetme.conf': Found
WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open
pseudo channel
  -- Playing 'conf-invalid'
  -- Playing 'conf-getconfno'
  -- Playing 'conf-getconfno'
Thanks



 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
   



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Active ISDN PCMCIA card

[Michael Manousos]

One more thing, just to be sure.
Are these (AVM B1, Fritz) Cardbus cards?
Nothing is mentioned on the web site.
Thanks,
Michael.
Michael Manousos wrote:
Thanks for the replies.
It seems that AVM B1 is the only active PCMCIA card that can be used
with Asterisk. The kernel supports this card, so I guess that the
driver can be built on non-x86 systems.
Regards,
Michael.


Olaf Menzel wrote:

On Friday 20 June 2003 13:28, Michael Manousos wrote:

Are there any suggestions for active ISDN CAPI PCMCIA cards
that are known to work with Asterisk?


You can try AVM B1 PCMCIA. This card is fully I4L compliant but AVM 
has developed a LINUX capi 2.0 stack. 
http://www.avm.de/en/products/hardware/active/B1_PCMCIA/index.html
The Linux Capi driver you find here:
ftp://ftp.avm.de/cardware/b1_pcm/linux/
Be aware that the Capi4Linux driver is distributed only as binary and 
especially prepared for Suse distributions. WIth some adaptations it 
should work with other distributions as well. Otherwise you should use 
I4L for this card. BTW. The Capi4Linux driver works also for the AVM 
Fritz which is much cheaper than the B1 device and supports full CAPI 
functionality such as G3 Fax.

regards

Olaf


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Ringing tones oh323

[Michael Manousos]

Jorge Cisneros wrote:
 
 
When i make a call using oh323 channels, how i can send a ringing sounds 
to indicate to the users that the call is in progress
This is generated by the IP phone. You don't have to do anything
special. In the case that the call has already been answered, and
you want to give back some kind of indication, you could
playback a prerecorded tone.
 
 
thanks
 


Michael.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Setting up the E100P

[Anton Yurchenko]

Martin Pycko wrote:

THat's not it.
in zapata.conf you *also* need to have
signalling=pri_cpe or pri_net
that did the work, I already had ( I guessed about this :) the pri_cpe 
option in zapata.conf.
so having in zaptel.conf:

bchan=1-15,17-31
dchan=16
and in zapata.conf:

signaling=pri_cpe

made E1 work with PRI signaling

Also my guess that Cisco`s switchtype primary-net5 ~= Asterisk EuroISDN 
am I right? This at least works with Seimens EDSW

I think we should have this added to documentation in 
/etc/asterisk/zapata.conf

thank a lot to Michael Blelicki and Martin Pycko!!

Still does anybody know what do all this stuff in zttool mean?

Thnks!!

 



--

Anton Yurchenko<[EMAIL PROTECTED]>
Digital Generation
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] App queue only + waiting call pickup

[Matteo Brancaleoni]

Hi.

Today I was asked about a function of asterisk.
That's what it should to:

a call arrives -> put it in a queue -> remain here ;)
Then, when someone wants to answer, just dial an extension
and the older call that's in the queue is picked up.
A sort of app_queueonly + app_pickolderqueuecall .

As far as I know asterisk doesn't support that, so
was wondering if someone encountered a similar request,
or is working on it.
I could start writing a new app, perhaps based on
app_queue , or is better an agi script ?

Or I'm missing a clever method to do it?

Matteo.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] TDM10B Configuration

[Sergio Serrano Revuelto]

Hi,
I try configure TDM10B to connect a fax. My problem is the next,
When fax direction is from out to *, fax sending goes well, but
when fax direction is from * to out, fac hangs up. I have connected a
phone to TDM10B and if I pickup phone I hear a strong tone and dtmf
tones are not recognize by TDM10B.

Must I configure some in zapata.conf? My zapata.conf for TDM10B is the
next:
group => 2
context=fax-ext
signalling=fxo_ks
callreturn=yes
busydetect=yes
busycount=10
immediate=no
rxgain = 0.8
txgain = 0.8
channel => 2


Any idea?

Thanks in advance,
srsergio

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] App queue only + waiting call pickup

[Adam Goryachev]

> Today I was asked about a function of asterisk.
> That's what it should to:
>
> a call arrives -> put it in a queue -> remain here ;)
> Then, when someone wants to answer, just dial an extension
> and the older call that's in the queue is picked up.
> A sort of app_queueonly + app_pickolderqueuecall .
>
> As far as I know asterisk doesn't support that, so
> was wondering if someone encountered a similar request,
> or is working on it.
> I could start writing a new app, perhaps based on
> app_queue , or is better an agi script ?
>
> Or I'm missing a clever method to do it?

Just setup a extra parking range of extensions, say 750 -799 (depends on how
many calls you might have queud at a time).
Then, just have another extension (say 800 to pull a call from the 'queue')
which just runs through from 750 to 799, the first call it finds is
connected through to the user.

Although, this doesn't acknowledge what happens when the call is answered by
an agent, you park a new call and it gets a lower number...

Personally, I'd like a feature similar to the app_queue but which doesn't
require the agent's phone to be off-hook all the time. This way, the agent's
phone can ring like normal and they can answer it. This better suits an
'office' environment where the 'agents' are not on the phone all day, but
might get 2 or 3 calls per hour (or even less)...

Regards,
Adam

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] App queue only + waiting call pickup

[Shaun Ewing]

> Personally, I'd like a feature similar to the app_queue but which doesn't
> require the agent's phone to be off-hook all the time. This way, the
agent's
> phone can ring like normal and they can answer it. This better suits an
> 'office' environment where the 'agents' are not on the phone all day, but
> might get 2 or 3 calls per hour (or even less)...

It's there.

eg (from queues.conf)
[sales]
music => moh-queue-announce
member => SIP/7011
member => SIP/7012
member => SIP/7013
member => Agent/17
member => Agent/18
member => Agent/19
context = callqueues-breakout

You can also put Zaptel (eg: Zap/1/whatever) in there.

Regards,
Shaun

> Regards,
> Adam
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] App queue only + waiting call pickup

[Brancaleoni Matteo]

Hi

Il mar, 2003-06-24 alle 14:32, Adam Goryachev ha scritto:

> > Today I was asked about a function of asterisk.
> > That's what it should to:
> >
> > a call arrives -> put it in a queue -> remain here ;)
> > Then, when someone wants to answer, just dial an extension
> > and the older call that's in the queue is picked up.
> > A sort of app_queueonly + app_pickolderqueuecall .
> >
> > As far as I know asterisk doesn't support that, so
> > was wondering if someone encountered a similar request,
> > or is working on it.
> > I could start writing a new app, perhaps based on
> > app_queue , or is better an agi script ?
> >
> > Or I'm missing a clever method to do it?
> 
> Just setup a extra parking range of extensions, say 750 -799 (depends on how
> many calls you might have queud at a time).
> Then, just have another extension (say 800 to pull a call from the 'queue')
> which just runs through from 750 to 799, the first call it finds is
> connected through to the user.

mmh, and if I free the first parking exten, like 750, then another
call arrives, is parked into 750... so I don't have a real FIFO...
and ..how can I pull a call from the parking lot, without dialling
the parked extension ?

> 
> Personally, I'd like a feature similar to the app_queue but which doesn't
> require the agent's phone to be off-hook all the time. This way, the agent's
> phone can ring like normal and they can answer it. This better suits an
> 'office' environment where the 'agents' are not on the phone all day, but
> might get 2 or 3 calls per hour (or even less)...

That should be accomplished with the standard app_queue,
just define members as channels (not agents) like Zap/xx or SIP/blah 
or anything you would use in a standard Dial argument,
and put incoming calls in the queue... then the member
tecnologies will start to ring ;)
That's what I use in standard office enviroments ;)


> 
> Regards,
> Adam

Matteo

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asteisk, sip & NAT

[Andrew Radke]

Hervé Thibaud wrote:

Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit :
...
 

I try to connect directly the both to fwd.pulver.com and now i have a
perfect sound but the question is perhaps links after opening session 
is only on the local networks with 10Mb/s.
Once i can (when i'll have an external user to call) i'll try.
   

It's like i thought, the sound is nasty with many blanks when i try a
connection on internet and ISDN bandwith on one channel 64kb/s is not
enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and
use a bandwith of 64kb/s so. 
Is anybody that has a free or a very cheap solution (it's to try
asterisk) to have an IP phone hardware or software with G.723.1 codec

Other thing, I would like to try X-Lite but i have pb with registration,
i don't know how to write settings, i have an error (for example) : 
File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau
" failed for '192,168,0,1'
I try many form of settings but didn't succeed. 

X-Lite support speex (SPX on the X-Lite screen). This is a very 
impressive codec that will even allow you to talk with someone over a 
_modem_ with a little bandwidth to spare for other stuff. Obviously the 
modem adds a degree of latency but it's still impressive. So with the 
latest technology, etc we've managed to get a _voice_ conversation to 
travel over a standard phone line. ;-) But seriously, it is impressive.

Regards,

Andrew Radke   ,-_|\
[EMAIL PROTECTED]   mobile: +61 412 798593  / \
Member, System Administrators Guild of Australia  \_,-._*
   o
"I didn't know it was impossible when I did it."
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] databases for billing

[Jim Friedeck]

Don't forget you mst have MySQL-dev package installed for the cdr_mysql 
app to be compiled when you build Asterisk. The Makefile looks for the 
MySQL includes and libraries. Good luck.

Jim Friedeck



Ing. Angel Gomez Garcia wrote:

hostname=localhost
dbname=asteriskcdrdb
password=
user=asteriskcdruser
carlos del mayor wrote:

I'm only asking for some examples of cdr_mysql.conf, nobody has done 
anything with cdr and mysql? If you think is better another DB,,, 
tell me, please!
thanks in advance
carlos

*/carlos del mayor <[EMAIL PROTECTED]>/* wrote:

can you be more explicit, please? or give me some examples?
please, i'm little lost!
thanks a lot
carlos
*/Martin Pycko <[EMAIL PROTECTED]>/* wrote:

cdr_mysql.conf

On Fri, 20 Jun 2003, carlos del mayor wrote:

> hi
> I want to do a database to save the cdr with a billing
finality. I've created the database in mysql (thanks for the
table and all that!) but I'm not sure of how to 'connect'
asterisk to that database in order to save there the cdr. Is
the cdr_mysql.conf what I have to config? Or must I do a
script, with the 'database' AGI commands?
> Any help would be so apreciated!
> Thanks a lot
> carlos
>
>
> -
> Yahoo! Sorteos
> Juega a la Lotería Primitiva sin salir de casa


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

*Ya! hoo! Sorteos*
*Juega a la Lotería Primitiva sin salir de casa*



*Yahoo! Sorteos*
*Juega a la Lotería Primitiva sin salir de casa* 
 





___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk hogging CPU resources

[Derek Beaumont]

Below is the portion of the ps information about Asterisk.  Note that
Asterisk started out at less than 5% CPU, and after 1075:35, it is now
at 74.6%.  The interfaces I am using are 2 X100P, and a TDM40B.  

>>I know there had been a bug some
>>time back that caused every asterisk thread to open handles on
>>/dev/zap/timer repeatedly and at some point my system had run out of
>>file handles to give out and performance started sucking. A CVS
upgrade
>>fixed that.

I upgraded asterisk yesterday, and this problem still occurs.  Also, I
am not sure what is meant by opening handles on /dev/zap/timer; I don't
have anything located at /dev/zap/timer.

USER   PID %CPU %MEM   VSZ  RSS TTY  STAT START   TIME COMMAND
root  9911 74.6  6.3 113296 7976 ?   RJun23 1075:35 asterisk
-vvvcg
root  9912  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9913  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9914  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9915  0.0  6.3 113296 7976 ?   SJun23   0:17 asterisk
-vvvcg
root  9916  0.0  4.0  7148 5176 ?SJun23   0:01
/usr/bin/mpg123 -
root  9917  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9918  0.0  0.8  3952 1076 ?SJun23   0:00
/usr/bin/mpg123 -
root  9919  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9920  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9921  0.7  6.3 113296 7976 ?   SJun23  10:07 asterisk
-vvvcg
root  9922  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9923  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9924  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9925  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9926  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg
root  9928  0.1  6.3 113296 7976 ?   SJun23   1:34 asterisk
-vvvcg
root  9929  0.0  6.3 113296 7976 ?   SJun23   0:00 asterisk
-vvvcg

Any help is always appreciated.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Monday, June 23, 2003 2:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk hogging CPU resources

What appears to be hogging CPU?  What interfaces are you running?

Mark

On Fri, 20 Jun 2003, Derek Beaumont wrote:

> Here's the problem:
>   I start asterisk, and it takes up around 3-4% of my CPU
> resources.
>   However, this number continues to climb over the hours until it
> is close to 100%.
>   Usually it takes around a day to climb up to approximately 95 or
> 96%
>
> Has anybody experienced the following problem before?
>
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] A question about cdr table

[Rafael Gonzalez Lomeña]

Hi,

  I have a little question:

  the calls from an 'oh323' terminal to another 'oh323' are they inserted
into the cdr table in MySQL?

   "CDR table" has only the calls, to or from a zap channel, but it doesn't
have any 'oh323' call. Any idea about this?



Regards.



Rafael Gonzalez Lomeña




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 911/Emergency calls + Caller ID

[Chris Witte]

You risk hanging up on your other 911 callers... but everything is 
always a tradeoff.

In my experience, the 911 dispatcher can (does) pin the call, so that 
even though the remote side hangs up, the line is not available for use 
again until the dispatcher releases it.   I'd expect this to mean that 
the proposed "hangup" would end up with the 911 operator transferred 
from caller->caller->caller if asterisk were configured to re-use a line 
for a new outbound 911 call...

chris.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] dynamic queue channels

[Benjamin Miller]

Title: Message



There 
may be some trickiness that can be done with "chan_local" as agents of the 
call queue.  However, a much more elegant way to do this would be to create 
an app_addagent and app_removeagent that allows the dynamic addition and removal 
of extensions from the agent pool for a given queue. addagent(${CHANNEL}, 
techsupport) or something like that.
Ben

  
  -Original Message-From: Paulo Mannheimer 
  [mailto:[EMAIL PROTECTED] Sent: Monday, June 23, 2003 6:36 
  PMTo: [EMAIL PROTECTED]Subject: 
  [Asterisk-Users] dynamic queue channels
  
  Hi, I’m trying to build a call 
  center application that allows attendants to come in the morning and dial a 
  certain extension to make their extension available. 
  
   
  I wouldn’t like to use the AgentLogin app because their line would need to stay 
  off-hook (is this correct?)
   
  Is there any SET channel status 
  command that would allow me to do something like 
  this?
   
  PauloHM
   

[Asterisk-Users] Chan Local Examples

[Matteo Brancaleoni]

There's anyone that's using chan_local that would
provide to the community some example on how to
use it or what's needed for ?

Just for knowledge.

Matteo.

-- 

Matteo Brancaleoni
Espia System Administrator
http://www.espia.it

-
This mail sent through IMP: http://horde.org/imp/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] "NoOp" gives an ringing indication ?

[Thomas Haeger]

Hi all,

i want lock Zap channels via global var FREE1

if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel

But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication !?

The "immediate" property in the zapat.conf is "yes"

[tel1]
exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten => s,2,Playback,gesperrt
exten => s,3,Hangup
exten => s,4,NoOp


exten => _X.,1,Dial,Modem/g1:${EXTEN}
exten => _X.,102,Hangup

What can i do ?



Thanks for help,

Thomas.



***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
***

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] PHP MySQL cdr interface?

[Marcus Adolfsson]

Title: Message



Before I 
"reinvent-the-wheel", does any one know of PHP based interface to the 
CDR table? If not, I will get started writing one.
 
Thanks,
 
Marcus 

[Asterisk-Users] parsing bug? (using PGSQL)

[Thomas Haeger]

Hi all again,

if i make a query with
...
exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT getdest('${EXTEN}')";
...

an error like

WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No application
'PGSQL,"Query resultid ${connid} 'SELECT getdest'
for extension (tel1, 00905888, 2)

occurs.

This looks like an parsing bug. As if the brackets be cut.

Can somebody help?

Thanks,

Thomas.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Conference calls on Pingtel Phones

[Andy Hester]

Has anyone been able to get conference calls to work on the Pingtel Phones?
I assume this feature works with their implementation, but connected to my
asterisk box it doesn't work.  The Pingtel phone thinks it is making a
second call, but asterisk never sees anything about a second call.  Any help
would be appreciated.

Sincerely,
Andy Hester
Consero

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 911/Emergency calls + Caller ID

[Dylan VanHerpen]

That would be the case if calls are dropped at random to clear the way 
for 911 calls. With some form of access control (NCOS, Calling Search 
Space/Partitions, priority levels) you would be able to drop the least 
important calls.

BTW, how are trunk restrictions managed right now? How can I specify 
which phones/extensions can make local, long distance or international 
calls? Can this be controlled by a time-of-day schedule, to change 
restrictions after regular business hours (cleaning crew calling LD)?

Dylan.

Chris Witte wrote:

You risk hanging up on your other 911 callers... but everything is 
always a tradeoff.

In my experience, the 911 dispatcher can (does) pin the call, so that 
even though the remote side hangs up, the line is not available for 
use again until the dispatcher releases it.   I'd expect this to mean 
that the proposed "hangup" would end up with the 911 operator 
transferred from caller->caller->caller if asterisk were configured to 
re-use a line for a new outbound 911 call...

chris.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] PHP MySQL cdr interface?

[Iain Stevenson]

Roy Sigurd Karlsbakk posted a php utility to calculate call costs to this 
list a while back.  I hacked it for my own use and you can have that if 
you'd like to improve it/make it general purpose.

 Iain



--On Tuesday, June 24, 2003 11:18 am -0400 Marcus Adolfsson 
<[EMAIL PROTECTED]> wrote:

Before I "reinvent-the-wheel", does any one know of PHP based interface
to the CDR table? If not, I will get started writing one.
Thanks,
Marcus




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk and Polycom

[Chris]

I am trying to configure Asterisk to use a Polycom 
SoundPoint IP 500 phone.  Does anyone know where I can get the software and 
configuration file for this phone.  I spoke to Polycom support and they say 
it's up to the SIP vendor to provide this.

Re: [Asterisk-Users] "NoOp" gives an ringing indication ?

[John Todd]

Where in here are you waiting for digits?  I don't see anything in 
here that allows input from the user.  You are getting the ring 
because the "s" extension is completed, and ${EXTEN} is set to 
something, and then you're jumping right to _X. which is finding a 
match and executing the Dial application.

You need a "Background" application in the "s" extension range 
somewhere that waits for digits.

JT


Hi all,

i want lock Zap channels via global var FREE1

if FREE1 = 1 then call should go on with nothing and waiting for digits to
go in _X.
Otherwise hangup the channel
But if the GotoIf goes to s|4 (NoOp) then comes a ringing indication !?

The "immediate" property in the zapat.conf is "yes"

[tel1]
exten => s,1,GotoIf($[${FREE1} = 1]?s|4:s|2)
exten => s,2,Playback,gesperrt
exten => s,3,Hangup
exten => s,4,NoOp
exten => _X.,1,Dial,Modem/g1:${EXTEN}
exten => _X.,102,Hangup
What can i do ?



Thanks for help,

Thomas.



***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email:  [EMAIL PROTECTED]
***
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] asterisk and passwords

[John Todd]

Currently there is no way to store SIP passwords in an encrypted 
form, and there is no user interface to change passwords that is 
built into the Asterisk system.

You will need to build your own user interface.

Note that you could, if you really, REALLY wanted to, build your own 
password encryption method using some of the tools in the ser 
toolkit.  An easier way to do it would be to allow users to change 
their passwords via an AGI script (perl?) that had an interface 
through a voice prompt menu within Asterisk.

JT


Hi.

Perhaps I just didn't found it, but there are two things I'm currently missing
in asterisk:
- -   The user passwords for SIP are written *plain ascii* in the
sip.conf... is there any possibility to store them encryptet?
- -   How can a user change his(her) SIP-Password? Is there any key in
the voicemail/agent-system which allows changing the SIP-Password?
The only thing I've found there is the possibility of changing the
voicemail password. Or exists somewhrer an web-based form (like the
SIPExpressRouter has) to do so?
Thanks for your help
 Rainer
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] parsing bug? (using PGSQL)

[Martin Pycko]

If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)

That should work

regards
Martin

On Tue, 24 Jun 2003, Thomas Haeger wrote:

> Hi all again,
>
> if i make a query with
> ...
> exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT getdest('${EXTEN}')";
> ...
>
> an error like
>
> WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No application
> 'PGSQL,"Query resultid ${connid} 'SELECT getdest'
> for extension (tel1, 00905888, 2)
>
> occurs.
>
> This looks like an parsing bug. As if the brackets be cut.
>
> Can somebody help?
>
> Thanks,
>
> Thomas.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] 911/Emergency calls + Caller ID

[Jon Pounder]

make a context for l/d dialing and include it for the phones / times of 
day, when it is actually supposed to be used, not otherwise.

At 09:52 AM 6/24/2003 -0600, you wrote:
That would be the case if calls are dropped at random to clear the way for 
911 calls. With some form of access control (NCOS, Calling Search 
Space/Partitions, priority levels) you would be able to drop the least 
important calls.

BTW, how are trunk restrictions managed right now? How can I specify which 
phones/extensions can make local, long distance or international calls? 
Can this be controlled by a time-of-day schedule, to change restrictions 
after regular business hours (cleaning crew calling LD)?

Dylan.

Chris Witte wrote:

You risk hanging up on your other 911 callers... but everything is 
always a tradeoff.
In my experience, the 911 dispatcher can (does) pin the call, so that 
even though the remote side hangs up, the line is not available for use 
again until the dispatcher releases it.   I'd expect this to mean that 
the proposed "hangup" would end up with the 911 operator transferred from 
caller->caller->caller if asterisk were configured to re-use a line for a 
new outbound 911 call...

chris.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] PHP MySQL cdr interface?

[WipeOut .]

I don't know of one but would be very interested in getting hold of one.. If you are 
going to realese it please let me know..

Thanks..

> Before I "reinvent-the-wheel", does any one know of PHP based interface
> to the CDR table? If not, I will get started writing one.
>  
> Thanks,
>  
> Marcus 

-- 
__
http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr

Powered by Outblaze
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update

[Surfer Dude]

TC,

Thank you very much for the pointer.  I wanted to acknowledge how valuable it was.  I was able to monitor the channel.  I did not know that tool exsited or what it did.  Now I do.

I actually do not know what an FSK tone is.  I gather, from google, that it is some audible tone between the first and send ring.  I was not able to to hear anything between the rings.  However, my caller ID stopped working in loopstart mode as well.  So... I have to dig deeper.

Thanks again,

Jason


On Friday, June 20, 2003, at 10:47  AM, TC wrote:

Callerid issue
1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone
 if yes then we beleive the CAC is passing fsk
    2) in chan_zap->ss_thread around line 4154 (current cvs)
    if you get to the callerid_feed at least once then
  if you get to chan_zap->ss_thread->callerid_get  around line 4163 (current cvs)
    does this parse fail
  else
    do you get check sumfails or sum fin else
    else
  zaptel is not detecting the ring/fsk correct
  might need to tunimng on ZT_RINGTRAILER
 else
    CAC is hooped

-Original Message-
From: Surfer Dude <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED] <[EMAIL PROTECTED]>
Date: June 20, 2003 10:24 AM
Subject: Re: [Asterisk-Users] Question :: groundstart and loopstart :: Update

Thanks for the responses.
 
Here is an update on my zap groundstart, loopstart, "disconnect supervision" woes.  Apparently, the groundstart mode in the CAC FXO module works on loopstart lines.  I still don't understand how and why.
 
I have loopstart lines.
 
The only way "disconnect supervision" works on my system is if I set the FXO module to groundstart.  However!  There is a big caveat.  In groundstart mode, incoming caller ID stops working!  This is something I am going to have to live with for now as disconnecting the line on hang-up is far more important than caller ID.
 
I have no idea if the caller ID problem is a problem with CAC FXO modules or the zap channel software or the fact that we are using groundstart signaling on a loopstart line.
 
I can confirm that this is the behavior with both CAC I and CAC II units as I have a CAC II unit and I spoke with Michael on this list who has a CAC I unit.  Both of us are in the same predicament.
 
Anyway, I hope others find this info useful.  If anyone has any other ideas on how I can get my caller ID to work I would be really excited about that.
 
 
Thanks,
Jason
 
PS: I have seen people mention that, when you have many lines, PRI is cheaper.  This is not our case at all.  San Francisco.  We have eight lines at the cost of $0.01 per line.  All we pay for really is the call time.
 

- Original Message -
From: Surfer Dude
To: [EMAIL PROTECTED]
Sent: Tuesday, June 17, 2003 4:05 PM
Subject: [Asterisk-Users] Question :: groundstart and loopstart

Hello Astrites,
 
I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines.  I had tried many configurations.  I had attached a toner to the line to see if I was getting this "disconnect supervision" signal after hang-up.  (toner has a line powered light in the off position)  It seemed that I was getting the signal, as:
 
    The light was steady on onhook
    When the line rang the light dimmed.
    When I picked up the line stayed dim (If I remember correctly)
    When the other end hung up the light stayed on
    Then turned off momentarily a few seconds later.  (I suspected this was "disconnect supervision")
 
Anyway asterisk never detected hang-up.  Everything else worked.  I was frustrated as hell.
 
After a few days of trying everything, I tried to set my CAC II FXO module to Groundstart and configured zaptel.conf and zapata.conf to groundstart signaling.
 
Voila!  Everything work!
 
The question is how does groundstart and loopstart work?  How was I able to have a somewhat working system using loopstart when it seems that the phone company signaling was groundstart?  Can I assume that my lines are indeed groundstart?
 
I am really happy that it works.  I can now begin the process of deployment with a much greater feeling of security.
 
Jason
 
 

AW: [Asterisk-Users] parsing bug? (using PGSQL)

[Thomas Haeger]

OK. I see.

This works.

Thank you,

Thomas.

-Ursprungliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Martin
Pycko
Gesendet: Dienstag, 24. Juni 2003 18:03
An: Asterisk User
Betreff: Re: [Asterisk-Users] parsing bug? (using PGSQL)


If you use brackets () then you need to call it like this
PGSQL(blabla(bla)bla)

That should work

regards
Martin

On Tue, 24 Jun 2003, Thomas Haeger wrote:

> Hi all again,
>
> if i make a query with
> ...
> exten => _X.,2,PGSQL,"Query resultid ${connid} SELECT
getdest('${EXTEN}')";
> ...
>
> an error like
>
> WARNING[32785]: File pbx.c, Line 1126 (pbx_extension_helper): No
application
> 'PGSQL,"Query resultid ${connid} 'SELECT getdest'
> for extension (tel1, 00905888, 2)
>
> occurs.
>
> This looks like an parsing bug. As if the brackets be cut.
>
> Can somebody help?
>
> Thanks,
>
> Thomas.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk and Polycom

[Benjamin Miller]

Title: Message



Yeah, 
we tested this.  Polycom was a real butt about it too.  They have 
sealed some agreements with vendors that sell their phones with their 
proprietary systems.  While Polycom admits the phone does "SIP" they will 
not disclose any info about configuring it and the vendors phone systems that 
configure it have system specific firmware and xml config files.  So, my 
take on the Polycom is that they say "SIP" but it is not really usable by 
anything but their partners phone systems.
So my 
advice is don't waste your time.
Ben

  
  -Original Message-From: Chris 
  [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 24, 2003 10:30 
  AMTo: [EMAIL PROTECTED]Subject: 
  [Asterisk-Users] Asterisk and Polycom
  I am trying to configure Asterisk to use a 
  Polycom SoundPoint IP 500 phone.  Does anyone know where I can get the 
  software and configuration file for this phone.  I spoke to Polycom 
  support and they say it's up to the SIP vendor to provide 
this.

Re: [Asterisk-Users] New Module app_perl

[Michael Baird]

Wow, sounds great, I'll give it a go, would be lovely if I can use it to
auth on called-from (from a mysql database), would be one way of making
the config realtime, without having to HUP asterisk on each config
change.

Regards
MIKE


> Remove the space behind .com, like so http://asterisk.650dialup.com/
> 
> Cheers, Dylan.
> 
> Uriel Carrasquilla wrote:
> 
> >For some reason the page cannot be found.
> > http://asterisk.650dialup.com
> >what does it do?
> >Uriel
> >
> >-Original Message-
> >From: [EMAIL PROTECTED]
> >[mailto:[EMAIL PROTECTED] Behalf Of Ask Bjørn
> >Hansen
> >Sent: Monday, June 23, 2003 5:12 AM
> >To: [EMAIL PROTECTED]
> >Subject: Re: [Asterisk-Users] New Module app_perl
> >
> >
> >
> >On Tuesday, Jun 17, 2003, at 20:43 America/Los_Angeles, Anthony
> >Minessale wrote:
> >
> >  
> >
> >>Here is a copy of the first release (comments appreciated)
> >> 
> >>http://asterisk.650dialup.com
> >>
> >>
> >
> >Although I haven't had time to play with it: very neat!
> >
> >
> >  - ask
> >
> >--
> >http://www.askbjoernhansen.com/
> >
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >  
> >
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk vs. system user accounts

[Dylan VanHerpen]

I've been scouring the archives for discussions on this:

Why doesn't Asterisk use system user accounts for each 
extension/mailbox? That would add the benefit of encrypted passwords, 
logical grouping, unified mail/voice mail accounts (using 
/var/spool/mail instead of /var/spool/asterisk). I can already imagine 
Festival reading my emails to me, HylaFAX faxing documents to me while 
I'm on the road :).

Dylan.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Reed Wade]

Because you haven't written and contributed that functionality yet.

(smiley face goes here)

That sounds pretty sweet. I'm wondering if LDAP might be the more
correct thing to use though.
-reed



At 10:50 AM 6/24/2003 -0600, you wrote:
I've been scouring the archives for discussions on this:

Why doesn't Asterisk use system user accounts for each extension/mailbox? 
That would add the benefit of encrypted passwords, logical grouping, 
unified mail/voice mail accounts (using /var/spool/mail instead of 
/var/spool/asterisk). I can already imagine Festival reading my emails to 
me, HylaFAX faxing documents to me while I'm on the road :).

Dylan.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Dylan VanHerpen]

I'm wondering if LDAP might be the more correct thing to use though.

Absolutely!

Reed Wade wrote:



Because you haven't written and contributed that functionality yet.

(smiley face goes here)

That sounds pretty sweet. I'm wondering if LDAP might be the more
correct thing to use though.
-reed



At 10:50 AM 6/24/2003 -0600, you wrote:

I've been scouring the archives for discussions on this:

Why doesn't Asterisk use system user accounts for each 
extension/mailbox? That would add the benefit of encrypted passwords, 
logical grouping, unified mail/voice mail accounts (using 
/var/spool/mail instead of /var/spool/asterisk). I can already 
imagine Festival reading my emails to me, HylaFAX faxing documents to 
me while I'm on the road :).

Dylan.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Working Clients for Linux?

[Moshe Yudkowsky]

All the clients that I'm aware of for IP telephony have drawbacks. Some 
won't work at all.

KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to 
dial tones during the middle of the call, so the demo that * comes with 
can't be run. Kphone (3.1, the latest) also has a habit of crashing if 
you do something even mildly stressful, such as hang up while Kphone is 
trying to connect.

LINPHONE -- Linphone does not work with ALSA, nor with ALSA's OSS 
emulation. I've used pre-packaged version of Linphone and compiled my 
own, using --enable-alsa to get it up and running. NOTE -- not all 
versions of ALSA let you do this! There's a silent failure to compile -- 
check config.log to make certain that the ASOUND_H is defined.

GNOPHONE -- Gnophone also does not work with ALSA, although I haven't 
yet tried to compile it from a tarball/CVS. With ALSA version 0.9.4, 
GnoPhone won't even give me a chance to configure the sound card. IIRC 
GnoPhone did work with 0.9.2 and possibly 0.9.3.

1. I invite comments from people who can get these clients to work with 
ALSA 0.9.4. I am working with the latest CVS of ALSA.

2. Are there other open-source clients that I've missed?

Regards,
 Moshe
P.S. I've fiddled with the CVS versions only enough to get things to 
compile. E.g., in linphone, configure.in must be changed to find 
 instead of . Well, ok, I did add 
-mfmath=sse to some of the Makefiles but that should be harmless enough...

--
 Moshe Yudkowsky * http://www.Disaggregate.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk ALSA module not working

[Moshe Yudkowsky]

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The 
module chan_alsa.so won't load even if the oss module, chan_oss.so, 
isn't loaded. There are no error messages.

I've been chasing ALSA/Asterisk/client problems in one form or another 
for some time now. In previous versions of Asterisk and ALSA -- i.e., 
last week -- I could load either chan_oss.so or chan_alsa.so; as of this 
morning, only chan_oss.so will load.

The symptoms are very straightforward: If chan_alsa.so loads, no 
subsequent module loads. An outward sign is that "show dialplan" will 
have just a few items and not the entire dialplan because pbx_config.so 
doesn't load, and important functions like "stop now" do not exist at 
the console.

There are no error messages, and just the ordinary debug messages. I 
haven't had any success deciphering the problem -- I'm still figuring 
out its extend and ramping up on new sections of Asterisk code.

I am using the latest (2003-06-24) CVS of asterisk.

Question: Is anyone using Asterisk's chan_alsa.so and ALSA version 0.9.4 
from a later CVS? I like to know if it's some problem with my local 
configuration or a bug I haven't found.

Related question: Are OSS and ALSA mutually incompatible? I would think 
so, but there's nothing in the documentation. If they should *not* be 
loaded simultaneously, I will submit a few lines to patch modules.conf.

--
 Moshe Yudkowsky * http://www.Disaggregate.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Patching Festival

[Derek Beaumont]

I just wanted to try out Festival, but I can't get it patched.
I'm thinking that there is something missing from the steps listed 
at http://www.marko.net/asterisk/archives/0209/0389.html.

>>tar xvzf festival-1.4.2-release.tar.gz 
>>patch -p0 >  (or wherever the patch is located)

When I run the patch command, I get the following:

[EMAIL PROTECTED] src]# patch -p0 http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] SIP REGISTER script

[John Todd]

Some of you have unusual SIP configurations, and this SIP perl script 
may be useful to get remote devices registering with your Asterisk or 
other SIP server.  Most Cisco routers, as an example, are too stupid 
to REGISTER, so this script would be required to dynamically register 
them with a remote server.  This may not be 100% applicable to 
Asterisk, since static registrations are possible, but who knows? 
Maybe someone will find this useful.  There is no password support on 
this version.  I haven't fooled around extensively with the script - 
feel free to modify and re-submit a better working copy.

If anyone wants to build the password-capable version of this, some 
tools can be found in the SER package (http://www.iptel.org/ser/) in 
the "utils/gen_ha1" directory - there is a password generator in 
there, along with C routines for WWW-Authorize responses.

Notes:

Replace "123.123.123.123" with the IP address of the remote SIP 
server that you're trying to REGISTER with.

Replace "sipdomain.company.com" with the domain you're using.  This 
may not matter much for some SIP servers, but others are fussy about 
it.

Change the "Expires: " value to whatever you think is useful.  It's 
measured in seconds.

Change "John Doe " to your contact info 
(probably something like "")

JT



Original script by Marian Durkovic

 cut here ---

#!/usr/bin/perl
use Socket;
# USAGE register { local IP address, Extension number, SIP contact address }
register("1.2.3.4", "9", "John Doe ");
sub register {
$local_ip = shift;
$ext_number = shift;
$contact = shift;
$proxy_ip = "123.123.123.123";
$tm = time();
$seq = $tm % 65536;
$MESG="REGISTER sip:sipdomain.company.com SIP/2.0
Via: SIP/2.0/UDP $local_ip:5060
From: 
To: 
Contact: $contact
Call-ID: [EMAIL PROTECTED]
CSeq: $seq REGISTER
Expires: 3700
Content-Length: 0
";

$proto = getprotobyname('udp');
socket(SOCKET, PF_INET, SOCK_DGRAM, $proto) ;
$iaddr = inet_aton("$local_ip");
$paddr = sockaddr_in(5060, $iaddr);
bind(SOCKET, $paddr)   ;
$port=5060;
$hisiaddr = inet_aton($proxy_ip)  ;
$hispaddr = sockaddr_in($port, $hisiaddr);
send(SOCKET, $MESG, 0,$hispaddr  ) || warn "send $host $!\n";

}

 cut here ---
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Patching Festival

[sjacobs]

What directory are you in while running this? If you are in the festival
directory try:
patch -p1  I just wanted to try out Festival, but I can't get it patched.
> I'm thinking that there is something missing from the steps listed
> at http://www.marko.net/asterisk/archives/0209/0389.html.
>
> >>tar xvzf festival-1.4.2-release.tar.gz
> >>patch -p0  >>(or wherever the patch is located)
>
> When I run the patch command, I get the following:
>
> [EMAIL PROTECTED] src]# patch -p0  can't find file to patch at input line 4
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |diff -u -r festival-1.4.2/lib/tts.scm
> festival-1.4.2-asterisk/lib/tts.scm
> |--- festival-1.4.2/lib/tts.scm Wed Jan  8 09:54:14 2003
> |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan  7 08:51:44 2003
> --
> File to patch:
>
>
> Which file am I supposed to patch?
>
> Thanks for your help.
> -Derek
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Asterisk ALSA module not working

[Wade Weppler]

I have exactly the same problem.  RedHat 8 w/latest Alsa.

-wade

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Moshe Yudkowsky
> Sent: Tuesday, June 24, 2003 2:17 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Asterisk ALSA module not working
> 
> Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The
> module chan_alsa.so won't load even if the oss module, chan_oss.so,
> isn't loaded. There are no error messages.
> 
> I've been chasing ALSA/Asterisk/client problems in one form or another
> for some time now. In previous versions of Asterisk and ALSA -- i.e.,
> last week -- I could load either chan_oss.so or chan_alsa.so; as of this
> morning, only chan_oss.so will load.
> 
> The symptoms are very straightforward: If chan_alsa.so loads, no
> subsequent module loads. An outward sign is that "show dialplan" will
> have just a few items and not the entire dialplan because pbx_config.so
> doesn't load, and important functions like "stop now" do not exist at
> the console.
> 
> There are no error messages, and just the ordinary debug messages. I
> haven't had any success deciphering the problem -- I'm still figuring
> out its extend and ramping up on new sections of Asterisk code.
> 
> I am using the latest (2003-06-24) CVS of asterisk.
> 
> Question: Is anyone using Asterisk's chan_alsa.so and ALSA version 0.9.4
> from a later CVS? I like to know if it's some problem with my local
> configuration or a bug I haven't found.
> 
> Related question: Are OSS and ALSA mutually incompatible? I would think
> so, but there's nothing in the documentation. If they should *not* be
> loaded simultaneously, I will submit a few lines to patch modules.conf.
> 
> --
>   Moshe Yudkowsky * http://www.Disaggregate.com
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Steven Critchfield]

LDAP is a _MUCH_ better solution to that problem than user accounts on a
machine. Even if you have a /bin/false shell, users could cause trouble
with your system. For security reasons you want to keep the number of
accounts low and the number of accounts with password protected access
even lower. So using the built-in passwords for a system would only make
you more likely to be rooted on a machine that your business will depend
upon.

After saying LDAP is a better choice than system users, I still wonder
why it is important to have users be able to change passwords here. In
small deployments, it isn't going to happen often enough to bother the
administrator. In office environments, you shouldn't have to modify
passwords. Only when you have people roving around should you be worried
about password changes to keep them secure. 

OF course you should also first try to see how hard LDAP is to get setup
and secured before trying to link it into asterisk.

On Tue, 2003-06-24 at 12:07, Reed Wade wrote:
> Because you haven't written and contributed that functionality yet.
> 
> (smiley face goes here)
> 
> That sounds pretty sweet. I'm wondering if LDAP might be the more
> correct thing to use though.
> 
> -reed
> 
> 
> 
> At 10:50 AM 6/24/2003 -0600, you wrote:
> >I've been scouring the archives for discussions on this:
> >
> >Why doesn't Asterisk use system user accounts for each extension/mailbox? 
> >That would add the benefit of encrypted passwords, logical grouping, 
> >unified mail/voice mail accounts (using /var/spool/mail instead of 
> >/var/spool/asterisk). I can already imagine Festival reading my emails to 
> >me, HylaFAX faxing documents to me while I'm on the road :).
> >
> >Dylan.
> >
> >
> >___
> >Asterisk-Users mailing list
> >[EMAIL PROTECTED]
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Patching Festival

[Wade Weppler]

Make sure the directory that festival is in is named "festival-1.4.2".  One
other user had the same problem.  The directory was called "festival".

-wade


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Derek Beaumont
> Sent: Tuesday, June 24, 2003 2:48 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Patching Festival
> 
> I just wanted to try out Festival, but I can't get it patched.
> I'm thinking that there is something missing from the steps listed
> at http://www.marko.net/asterisk/archives/0209/0389.html.
> 
> >>tar xvzf festival-1.4.2-release.tar.gz
> >>patch -p0  >>(or wherever the patch is located)
> 
> When I run the patch command, I get the following:
> 
> [EMAIL PROTECTED] src]# patch -p0  can't find file to patch at input line 4
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |diff -u -r festival-1.4.2/lib/tts.scm
> festival-1.4.2-asterisk/lib/tts.scm
> |--- festival-1.4.2/lib/tts.scm Wed Jan  8 09:54:14 2003
> |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan  7 08:51:44 2003
> --
> File to patch:
> 
> 
> Which file am I supposed to patch?
> 
> Thanks for your help.
> -Derek
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Patching Festival

[Steven Critchfield]

cd into the festival src directory, then use -p1 instead of -p0. This
will strip the first level directory off of each patch part since it
will then be matched against the current directory you are in.

On Tue, 2003-06-24 at 13:47, Derek Beaumont wrote:
> I just wanted to try out Festival, but I can't get it patched.
> I'm thinking that there is something missing from the steps listed 
> at http://www.marko.net/asterisk/archives/0209/0389.html.
> 
> >>tar xvzf festival-1.4.2-release.tar.gz 
> >>patch -p0  >>(or wherever the patch is located)
> 
> When I run the patch command, I get the following:
> 
> [EMAIL PROTECTED] src]# patch -p0  can't find file to patch at input line 4
> Perhaps you used the wrong -p or --strip option?
> The text leading up to this was:
> --
> |diff -u -r festival-1.4.2/lib/tts.scm
> festival-1.4.2-asterisk/lib/tts.scm
> |--- festival-1.4.2/lib/tts.scm Wed Jan  8 09:54:14 2003
> |+++ festival-1.4.2-asterisk/lib/tts.scmTue Jan  7 08:51:44 2003
> --
> File to patch:
> 
> 
> Which file am I supposed to patch?
> 
> Thanks for your help.
> -Derek
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Telephony Portal w/ Asterisk

[destan]

I was planning on setting up a web site/service like Net2Phone or maybe FWD,
to give VoIP service. I think this can be setup using * pbx, by giving every
user an extension and so on. Maybe I have to code a web interface for 
registration
and profile editing, but other than that, softphones would automatically
register with * anyway right? Should I use * or try something like SER? 
Anybody
done anything like this with asterisk? 

ps: BTW, thanks for the answers for my log question.
Umut
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Analog 2x8

[Richard Scobie]

Is anyone on the list running an Asterisk system with 2 x  X100P and 2 x 
TDM40 (4 port) cards? I am interested in your hardware setup.

Regards,

Richard

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Dylan VanHerpen]

After saying LDAP is a better choice than system users, I still wonder
why it is important to have users be able to change passwords here.
It would greatly simplify unified messaging: one account, all your messages (email, voice, fax) in one mailbox.



Steven Critchfield wrote:

LDAP is a _MUCH_ better solution to that problem than user accounts on a
machine. Even if you have a /bin/false shell, users could cause trouble
with your system. For security reasons you want to keep the number of
accounts low and the number of accounts with password protected access
even lower. So using the built-in passwords for a system would only make
you more likely to be rooted on a machine that your business will depend
upon.
After saying LDAP is a better choice than system users, I still wonder
why it is important to have users be able to change passwords here. In
small deployments, it isn't going to happen often enough to bother the
administrator. In office environments, you shouldn't have to modify
passwords. Only when you have people roving around should you be worried
about password changes to keep them secure. 

OF course you should also first try to see how hard LDAP is to get setup
and secured before trying to link it into asterisk.
On Tue, 2003-06-24 at 12:07, Reed Wade wrote:
 

Because you haven't written and contributed that functionality yet.

(smiley face goes here)

That sounds pretty sweet. I'm wondering if LDAP might be the more
correct thing to use though.
-reed



At 10:50 AM 6/24/2003 -0600, you wrote:
   

I've been scouring the archives for discussions on this:

Why doesn't Asterisk use system user accounts for each extension/mailbox? 
That would add the benefit of encrypted passwords, logical grouping, 
unified mail/voice mail accounts (using /var/spool/mail instead of 
/var/spool/asterisk). I can already imagine Festival reading my emails to 
me, HylaFAX faxing documents to me while I'm on the road :).

Dylan.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
   



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Steven Critchfield]

On Tue, 2003-06-24 at 15:47, Dylan VanHerpen wrote:
> After saying LDAP is a better choice than system users, I still wonder
> why it is important to have users be able to change passwords here.
> 
> It would greatly simplify unified messaging: one account, all your
> messages (email, voice, fax) in one mailbox.

This doesn't mean there needs to be a real account on the system. Exim
can deliver to non login accounts, Cyrus IMAP doesn't need real user
accounts, and I forget the pop3 daemon that works the same way. Hylafax
doesn't need a user to send you the faxes. So again, you only need some
simple way of keeping user/password mappings straight. LDAP would be
okay, if it was easy enough to setup and use. Not to mention for your
SIP users, LDAP can be used as a phone directory. I hope this helps you
move forward with something usable and secure. 
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Distinctive Ring Macro Example

[Eric Wieling]

I use the following macro for my extensions.  It only works with Zap
channels and assumes that any Caller*ID number that is 4 digits is an
internal call and all other calls are external calls.

Use like this:  exten => 1234,1,Macro(std-exten,Zap/4,20)


[macro-std-exten]
;
; Caller*ID is 4 digits (internal call)
;
exten => s/_,1,Dial(${ARG1}r2,${ARG2})
exten => s/_,2,Voicemail(u${MACRO_EXTEN})
exten => s/_,3,Hangup
exten => s/_,102,Voicemail(b${MACRO_EXTEN})
exten => s/_,103,Hangup
;
; Caller*ID is not 4 digits (external call)
;
exten => s,1,Dial(${ARG1},${ARG2})
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup


--Eric
-- 
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Distinctive Ring Macro Example

[John Laur]

Cool trick!

You could simplify this:

[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten => s/_,1,Dial(${ARG1}r2,${ARG2})

; Caller*ID is not 4 digits (external call)
exten => s,1,Dial(${ARG1},${ARG2})

; Both of the above lines go here next
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup


For those of you running Cisco 7960's and using the ALERT_INFO stuff,
You can use this version of the same thing. (I am now using this in my
config thanks to Eric's example):

[macro-std-exten]
; Caller*ID is 4 digits (internal call)
exten => s/_,1,SetVar(ALERT_INFO=1)

; Caller*ID is not 4 digits (external call)
exten => s,1,NoOp

; Both of the above lines go here next
exten => s,2,Dial(${ARG1},${ARG2})
exten => s,3,Voicemail(u${MACRO_EXTEN})
exten => s,4,Hangup
exten => s,103,Voicemail(b${MACRO_EXTEN})
exten => s,104,Hangup


John

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Eric Wieling
> Sent: Tuesday, June 24, 2003 4:38 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Distinctive Ring Macro Example
> 
> I use the following macro for my extensions.  It only works with Zap
> channels and assumes that any Caller*ID number that is 4 digits is an
> internal call and all other calls are external calls.
> 
> Use like this:  exten => 1234,1,Macro(std-exten,Zap/4,20)
> 
> 
> [macro-std-exten]
> ;
> ; Caller*ID is 4 digits (internal call)
> ;
> exten => s/_,1,Dial(${ARG1}r2,${ARG2})
> exten => s/_,2,Voicemail(u${MACRO_EXTEN})
> exten => s/_,3,Hangup
> exten => s/_,102,Voicemail(b${MACRO_EXTEN})
> exten => s/_,103,Hangup
> ;
> ; Caller*ID is not 4 digits (external call)
> ;
> exten => s,1,Dial(${ARG1},${ARG2})
> exten => s,2,Voicemail(u${MACRO_EXTEN})
> exten => s,3,Hangup
> exten => s,102,Voicemail(b${MACRO_EXTEN})
> exten => s,103,Hangup
> 
> 
> --Eric
> --
> BTEL Consulting
> 850-484-4535 x2111 (Office)
> 504-595-3916 x2111 (Experimental)
> 877-552-0838 (Backup Phone)
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Compiling Asterisk under Yellow Dog

[Serge Mankovski]

Hi,
I am trying to compile Asterisk under Yellow Dog 3.0 distributionn.
I am getting an  error
gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm
/usr/bin/ld: cannot find -lgsm
May be I need packages that my distribution does not include?
What do I need to download to get it compiled?
Thanks
Serge
_
The new MSN 8: smart spam protection and 2 months FREE*  
http://join.msn.com/?page=features/junkmail

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Anthony Wood]

No-one has mentioned PAM yet.  (pluggable authentication modules).

If you implement PAM in Asterisk, then you have LDAP/passwd shadow
windows etc. in one step.

Maybe the phone numbers in /etc/passwd will get used!

cheers,
Woody

On Tue, Jun 24, 2003 at 04:17:15PM -0500, Steven Critchfield wrote:
> On Tue, 2003-06-24 at 15:47, Dylan VanHerpen wrote:
> > After saying LDAP is a better choice than system users, I still wonder
> > why it is important to have users be able to change passwords here.
> > 
> > It would greatly simplify unified messaging: one account, all your
> > messages (email, voice, fax) in one mailbox.
> 
> This doesn't mean there needs to be a real account on the system. Exim
> can deliver to non login accounts, Cyrus IMAP doesn't need real user
> accounts, and I forget the pop3 daemon that works the same way. Hylafax
> doesn't need a user to send you the faxes. So again, you only need some
> simple way of keeping user/password mappings straight. LDAP would be
> okay, if it was easy enough to setup and use. Not to mention for your
> SIP users, LDAP can be used as a phone directory. I hope this helps you
> move forward with something usable and secure. 
> -- 
> Steven Critchfield  <[EMAIL PROTECTED]>
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Woody
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Compiling Asterisk under Yellow Dog

[Steven Critchfield]

On Tue, 2003-06-24 at 17:59, Serge Mankovski wrote:
> Hi,
> I am trying to compile Asterisk under Yellow Dog 3.0 distributionn.
> I am getting an  error
> gcc -shared -Xlinker -x -o codec_gsm.so codec_gsm.o -lgsm
> /usr/bin/ld: cannot find -lgsm
> 
> May be I need packages that my distribution does not include?
> What do I need to download to get it compiled?

Just a bit of education for you so you can help yourself later, -l flags
to gcc are to link against libraries and the name of the library is lib
+ what ever is appended to the -l flag. It appears you have the headers
for libgsm since it didn't complain and die on lack of headers, but it
appears that gcc and ld are unable to find libgsm. It might be that it
is installed to a directory that ld needs to be made aware of in
/etc/ld.so.config.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk SIP-to-SIP proxy

[Witold P. Krecicki]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a 
proxy, but when connecting SIP to SIP it works only as 'SIP registrar' 
forwarding SIP requests to client. Is it possible to make Asterisk work as a 
'proxy' so that any incoming calls would be ade to Asterisk and then 
internally forwarded to receiver? (for eg. when Asterisk is on host that is 
doing masquerade for internal network and receiver/caller is in this 
masqueraded network)
- -- 
Witold Piotr Krecicki (adasi) adasi [at] culm.net
GPG key: 7AE20871
http://www.culm.net
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.1 (GNU/Linux)

iD8DBQE++N659le3g3riCHERAom9AKCF460/VHVaVRBfCgRANvoJz7BjxACghbkf
S1LWBrHtnR661pQDQBls/MU=
=dSDZ
-END PGP SIGNATURE-


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk SIP-to-SIP proxy

[Steven Critchfield]

On Tue, 2003-06-24 at 18:28, Witold P. Krecicki wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a 
> proxy, but when connecting SIP to SIP it works only as 'SIP registrar' 
> forwarding SIP requests to client. Is it possible to make Asterisk work as a 
> 'proxy' so that any incoming calls would be ade to Asterisk and then 
> internally forwarded to receiver? (for eg. when Asterisk is on host that is 
> doing masquerade for internal network and receiver/caller is in this 
> masqueraded network)

Use the reinvite=no or canreinvite=no options. Search list for more
detail on these options.
-- 
Steven Critchfield  <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Analog 2x8

[Brian J. Schrock]

I am ...

Antec SL2600 case (The bronze one, I think the part number is a little
off)
KT4VL Motherboard
AMD Duron 950
Kingston DDR266/333? 128 MB DRAM
KNE110TX Ethernet Card
Linksys WMP11 Wireless card HostAP
Maxtor 80G HDD
One Case Fan
2x X100P
2x TDM40P


On Tue, 2003-06-24 at 16:51, Richard Scobie wrote:
> Is anyone on the list running an Asterisk system with 2 x  X100P and 2 x 
> TDM40 (4 port) cards? I am interested in your hardware setup.
> 
> Regards,
> 
> Richard
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Dual T400P, SMP, performance issues

[Alex Zarubin]

Title: RE: [Asterisk-Users] Dual T400P, SMP, performance issues





Mark & Oliver,


It is too early to say, but the picture is different now. Our dual CPU,
dual T400P box is up for 4 days, under the load of 10 - 100 simultaneous
PRI -> SIP calls. We installed 2.4.21 #2 SMP (it was still freezing after
that) and, what I think made the difference, recompiled zaptel-libpri-asterisk
with gcc 3.3.


The problem, on the way, was that asterisk wouldn't start after that. It was
crashing while loading mp3 and lpc10 codecs. We put 'noload' for these two
into modules.conf - temporary solution, of course.


There are problems, still, with multiple connections at the same time. Windows
to the box get frozen for a sec, D-channel error messages. The following
messages are dumped into /var/log/messages. What do you think?


Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: wait_on_irq, CPU 1:
Jun 24 18:23:25 mspgate03 kernel: irq:  1 [ 0 0 1 0 ]
Jun 24 18:23:25 mspgate03 kernel: bh:   0 [ 0 0 0 0 ]
Jun 24 18:23:25 mspgate03 kernel: Stack dumps:
Jun 24 18:23:25 mspgate03 kernel: CPU 0:0200 036f 00e14603
1802 0310 6647 008e0200 4803
Jun 24 18:23:25 mspgate03 kernel:    0078 001ffa02 5b490300
0600 01c7 074e0308 1afe 01c74d03
Jun 24 18:23:25 mspgate03 kernel:    2302 d708 e101
0900 01d7 f5030001 0423 09300207
Jun 24 18:23:25 mspgate03 kernel: Call Trace:    []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   [] []
[]
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 2:  
    
Jun 24 18:23:25 mspgate03 kernel:      
    
Jun 24 18:23:25 mspgate03 kernel:      
    
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 3:0070 cce30002 0cd8
08fa 6953 656c706d 6c616e41 73697379
Jun 24 18:23:25 mspgate03 kernel:    0009a700 46534c00 65746e69
6c6f7072 32657461 6e655f61 0a810063 6953
Jun 24 18:23:25 mspgate03 kernel:    656c706d 65746e49 6c6f7072
4c657461 39004653 530b 6c706d69 66736c65
Jun 24 18:23:25 mspgate03 kernel: Call Trace:
Jun 24 18:23:25 mspgate03 kernel:
Jun 24 18:23:25 mspgate03 kernel: CPU 1:e14d5eac c025c896 0001
0001  0001 c010a7c2 c025c8ab
Jun 24 18:23:25 mspgate03 kernel:     f2d92124 e14d5f00
c0191104 0500 1805 00bf 8a01
Jun 24 18:23:25 mspgate03 kernel:    7f1c0300 01000415 1a131100
170f1200  e14d4000  
Jun 24 18:23:25 mspgate03 kernel: Call Trace:    []
[] [] [] []
Jun 24 18:23:25 mspgate03 kernel:   []
Jun 24 18:23:25 mspgate03 kernel:


Thank you.
Alex Zarubin


-Original Message-
From: The Traveller [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, June 17, 2003 3:10 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dual T400P, SMP, performance issues



On Tue, Jun 17, 2003 at 20:54:39 +0200, The Traveller wrote:
> 
> BTW: As I reported in my previous mail to the list, I've now installed kernel
> 2.4.21-rc2 with ACPI-patch on the box with the E100P.  I've been trying
> very hard to reproduce a freeze with this kernel, but haven't succeeded yet.
[...]


Ok, it crashed again, so that wasn't it either.  What I did to trigger
it was using the auto-dialer to loop as many calls to app_datetime out
and then back over the same E-1 as it would take, queueing the calls
to "/var/spool/asterisk/outgoing/" 14 at a time.  It froze at the first
attempt.  The "good" news is that it produced a visible kernel-panic.
this time.  My guess is that you only don't see it if the console
screensaver has already come on while it happens.


It read something like "Unable to handle kernel paging request" and
happened in the swapper-task.  As usual, it dumped a lot of numbers on the
screen, which I didn't want to write down.


Mark: If you want my help in debugging this, I'll hook it up to a
serial console, trigger the crash and provide you with the exact
panic, together with the ksyms and modules-info to trace it.




    Grtz,


   Oliver
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Working Clients for Linux?

[Jeremy McNamara]

GnomeMeeting
ohphone -- not a  gui, but functional with a few tweaks (esp if 
openh.323 gets updated to deal with Zaptel)

Jeremy McNamara

Moshe Yudkowsky wrote:

All the clients that I'm aware of for IP telephony have drawbacks. 
Some won't work at all.

KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad 
to dial tones during the middle of the call, so the demo that * comes 
with can't be run. Kphone (3.1, the latest) also has a habit of 
crashing if you do something even mildly stressful, such as hang up 
while Kphone is trying to connect.

LINPHONE -- Linphone does not work with ALSA, nor with ALSA's OSS 
emulation. I've used pre-packaged version of Linphone and compiled my 
own, using --enable-alsa to get it up and running. NOTE -- not all 
versions of ALSA let you do this! There's a silent failure to compile 
-- check config.log to make certain that the ASOUND_H is defined.

GNOPHONE -- Gnophone also does not work with ALSA, although I haven't 
yet tried to compile it from a tarball/CVS. With ALSA version 0.9.4, 
GnoPhone won't even give me a chance to configure the sound card. IIRC 
GnoPhone did work with 0.9.2 and possibly 0.9.3.

1. I invite comments from people who can get these clients to work 
with ALSA 0.9.4. I am working with the latest CVS of ALSA.

2. Are there other open-source clients that I've missed?

Regards,
 Moshe
P.S. I've fiddled with the CVS versions only enough to get things to 
compile. E.g., in linphone, configure.in must be changed to find 
 instead of . Well, ok, I did add 
-mfmath=sse to some of the Makefiles but that should be harmless 
enough...




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problems with # and extensions.

[Eric Wieling]

I get the following message when I dial 74#.  Does anyone have any ideas
on what might be going on?  If I don't require numbers to be terminated
with # everything works as expected, but you have to wait for the digit
timeout, of course.

MESSEGE:

DEBUG[1150520624]: File pbx.c, Line 1683 (ast_pbx_run): Oooh, got
something to jump out with ('#')!
-- Invalid extension '#' in context 'speeddial' on SIP/2111-076f

CONFIG:

[default]
exten => 74#,1,Goto(speeddial,s,1)

[speeddial]

exten => s,1,Background(${MYSOUNDS}/enterspeedcode)

exten => 1#,1,Background(${MYSOUNDS}/speedhelp)
exten => 1#,2,Goto(s,1)

exten => _[2-9]#,1,SetVar(SPEEDCODE=${EXTEN})
exten => _[2-9]#,2,Background(${MYSOUNDS}/enterspeednumber)

exten => _X.#,1,SetVar(SPEEDNUMBER=${EXTEN})
exten => _X.#,2,Background(${MYSOUNDS}/speedcode)
exten => _X.#,3,SayDigits(${SPEEDCODE})
exten => _X.#,4,Background(${MYSOUNDS}/assignedto)
exten => _X.#,5,SayDigits(${SPEEDNUMBER})


-- 
BTEL Consulting
850-484-4535 x2111 (Office)
504-595-3916 x2111 (Experimental)
877-552-0838 (Backup Phone)

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Which type of lines to get from the Analog PBX??

[surajee]

hi,
I have the following scenario,
an Asterisk serveranolog pbx
I hav a channel bank, it accepts 30 analog lines coming from the analog pbx,the other end of the channel bank is the asterisk server with E100P card.I am confused as to which type of lines to get from the analog pbx.
I think its suffient to have normal analog extensions, which they are alreadyhaving in there pbx, plugged into analog phones.
When i contacted the pbx guys to do the upgrade for me(upgrade by 30 chs), they ask me whether, i need E&M lines or not.I am very sorry for asking this, if these are general knowledge stuff.I know that normal analog lines work with the loop current changes, and i found out that, In E&M (Ear and Mouth) line signalling the speech circuit (or channel) has an associated E-wire and M-wire for signalling. 
Can anybody pls tell me as to which type of lines should i get from the analog pbx,whether normal analog lines or E&M lines?
Thanx in advance,
Surajee

--This mail sent through OmniBIS.com--

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Tor2 modprobe problem

[asterisk]

I just received my dev kit and installed the hardware.  I followed the
directions Digium sends out to the letter and when I ran modprobe tor2 I
got the following error:

/lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters,
including invalid IO or IRQ parameters.
  You may find more information in syslog or the output from dmesg
/lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod
/lib/modules/2.4.18-19.7.x/misc/tor2.o failed
/lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed

What have I missed?

-- 
Ryan V. Howerton


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] chan_oh323.c Segmentation fault during Openphone/Gnomemeetingconnect during module loading...

[Jordan Peterson]

My apologies if this question has been answered previously.  However, I
found that it was nearly impossible to search and find since anything
can cause a segmentation fault.

Problem.  When Asterisk is booting up the h323 modules and a client
tries to connect before Asterisk/h323 is finished booting, the program
seg faults out and doesn't load.  I thought about putting this into the
inittab, but wouldn't that be more of a pain to manage from the initttab
(respawn, etc)?

Other than that, it boots up fine and works with h323 clients just
fine..Can you imagine the hell a person would go through trying to boot
up asterisk once it becomes more popular to use by clients?  For now, it
works fine because i am the only person who is connecting to it.. haha..

Is there a simple fix to this, or is everyone having this problem?

Thanks


-- 
Jordan


In a world without windows, who needs gates?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tor2 modprobe problem

[asterisk]

OK...I think I have found that answer to one problem, but now more have
cropped up.  Apparently I was using the wrong module.  What modules do I
need to load?  Are there any parameters that need to be passed?  Where can
I find more information about my cards.  I assumed that there would be
more information with the cards than I received.

Ryan

> I just received my dev kit and installed the hardware.  I followed the
> directions Digium sends out to the letter and when I ran modprobe tor2 I
> got the following error:
>
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device
> Hint: insmod errors can be caused by incorrect module parameters,
> including invalid IO or IRQ parameters.
>   You may find more information in syslog or the output from dmesg
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod
> /lib/modules/2.4.18-19.7.x/misc/tor2.o failed
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed
>
> What have I missed?
>
> --
> Ryan V. Howerton
>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tor2 modprobe problem

[Steven Critchfield]

On Tue, 2003-06-24 at 22:19, [EMAIL PROTECTED] wrote:
> I just received my dev kit and installed the hardware.  I followed the
> directions Digium sends out to the letter and when I ran modprobe tor2 I
> got the following error:
> 
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device
> Hint: insmod errors can be caused by incorrect module parameters,
> including invalid IO or IRQ parameters.
>   You may find more information in syslog or the output from dmesg
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod
> /lib/modules/2.4.18-19.7.x/misc/tor2.o failed
> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed
> 
> What have I missed?

This would be because your card was not detected. 

Since I haven't kept up with what was in the dev kit, do you have the 4
port T1 card? Tor2 is for the 4 port T1/E1 cards. If it is a single port
T1 card, you need to use the wct1xxp module. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Asterisk vs. system user accounts

[Rainer Jochem]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi.

I think LDAP (or perhaps even better: an generic ODBC interface) is really the 
way you want:
in a large environment (enterprise, university, ...) you often already have a
central database of users. If you now want to add an * system, nobody wants
to create thousands of user accounts on the asterisk machine. Additionaly,
if you use * with more than one servers (for redundancy, load balancing,
 ), it is much easier to let all the * systems use an central LDAP server 
for authentication rather to create system users on every single machine.

It is also easier to provide an interface to the useres for letting them
change their passwords (which is in my opinion also a very important
security issue) if there is one central database.

Greetings
 Rainer

- -- 
http://graphics.cs.uni-sb.de/~rainer/
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.0.6 (GNU/Linux)
Comment: For info see http://www.gnupg.org

iD8DBQE++TqeH/5+6U9F1qYRAsS4AJ9SfDEe7MEU6FmINWOUwfA2ZFP8fwCeIEjF
53V0FBfYx8Uj04j3BQSYSZc=
=u2LG
-END PGP SIGNATURE-

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Tor2 modprobe problem

[asterisk]

Here is what I have found.  I could not get the modules loaded.  I took
the TDM400P out of my machine moved it to another PCI slot and restarted. 
The modules loaded with no problems and asterisk runs just fine.

Ryan

> On Tue, 2003-06-24 at 22:19, [EMAIL PROTECTED] wrote:
>> I just received my dev kit and installed the hardware.  I followed the
>> directions Digium sends out to the letter and when I ran modprobe tor2
>> I got the following error:
>>
>> /lib/modules/2.4.18-19.7.x/misc/tor2.o: init_module: No such device
>> Hint: insmod errors can be caused by incorrect module parameters,
>> including invalid IO or IRQ parameters.
>>   You may find more information in syslog or the output from dmesg
>> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod
>> /lib/modules/2.4.18-19.7.x/misc/tor2.o failed
>> /lib/modules/2.4.18-19.7.x/misc/tor2.o: insmod tor2 failed
>>
>> What have I missed?
>
> This would be because your card was not detected.
>
> Since I haven't kept up with what was in the dev kit, do you have the 4
> port T1 card? Tor2 is for the 4 port T1/E1 cards. If it is a single port
> T1 card, you need to use the wct1xxp module.
> --
> Steven Critchfield <[EMAIL PROTECTED]>
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users