date:20040203

show application dial from asterisk cli:

snip

  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'r' -- indicate ringing to the calling party, pass no audio until
answered.
  'm' -- provide hold music to the calling party until answered.
  'H' -- allow caller to hang up by hitting *.
  'C' -- reset call detail record for this call.
  'P[(x)]' -- privacy mode, using 'x' as database if provided.
  'g' -- goes on in context if the destination channel hangs up
  'A(x)' -- play an announcement to the called party, using x as
file

see last param ...

Matteo.



-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Playing announcement to called user prior toConfirmation

I wish 'A(x)' was available with AgentCallBackLogin!! :(

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Brancaleoni
Sent: Tuesday, February 03, 2004 6:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Playing announcement to called user prior
toConfirmation

show application dial from asterisk cli:

snip

  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'r' -- indicate ringing to the calling party, pass no audio until
answered.
  'm' -- provide hold music to the calling party until answered.
  'H' -- allow caller to hang up by hitting *.
  'C' -- reset call detail record for this call.
  'P[(x)]' -- privacy mode, using 'x' as database if provided.
  'g' -- goes on in context if the destination channel hangs up
  'A(x)' -- play an announcement to the called party, using x as
file

see last param ...

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [Asterisk-Users] cisco 7912 voicemail/dnd issue

2004-02-03 Thread Swen Veckes


 Having exactly the same problem with 7905. In addition it doesn't
 seem you can
 disable (at least on my sip fw release) the
 redirect-to-vm-on-busy feature.


Yes, that's right, only the value for no answer can be changed (set to
high value == disable).

 Whenever the phone has the VM number configured (and people like the
 messages button, sadly) it sends out 302 messages for the vm
 number when
 it's busy, regardless of any other config setting, which is
 _very_ bad for
 queues and acd.

 I had to do a quick modification to chan_sip denying redirects to
 the (magic
 hardcoded) vm number.

 Hope for a new release where you can either set the vm, vm-listen
 separately,
 or at least disable the redirect feature.

 lele

Actually I think of a soultion like changeing the VM app to be canceld when
dialing
* and one can enter the vm and pin to access his messages otherwhise just
leave
a message. So I don't need to deal with different VMNo. for one user.
But no glue how to make it ;(


:wq swen

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Playing announcement to called user prior toConfirmation

Does anyone know if this feature is actually implemented? I just tried
it with a Dial statement of mine and it doesn't play any file.  Doesn't
report any errors, and I'm sure the file exists.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew B
Marlowe
Sent: Tuesday, February 03, 2004 6:55 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Playing announcement to called user prior
toConfirmation

I wish 'A(x)' was available with AgentCallBackLogin!! :(

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matteo
Brancaleoni
Sent: Tuesday, February 03, 2004 6:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Playing announcement to called user prior
toConfirmation

show application dial from asterisk cli:

snip

  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'r' -- indicate ringing to the calling party, pass no audio until
answered.
  'm' -- provide hold music to the calling party until answered.
  'H' -- allow caller to hang up by hitting *.
  'C' -- reset call detail record for this call.
  'P[(x)]' -- privacy mode, using 'x' as database if provided.
  'g' -- goes on in context if the destination channel hangs up
  'A(x)' -- play an announcement to the called party, using x as
file

see last param ...

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation

2004-02-03 Thread reseaux

Dear Matthew
yes it work great A(playmex) where playmex is gsm file in sound dir.. i have 
made some simple hack to app_dial.c to have a new option B(playmex) with it i 
can play a mex to the caller when the call is connected i use it to play a 
dtmf code...
Thanks in advance
Dimitri

PS: if someone think is good option a send it

On Tuesday 03 February 2004 12:21, Matthew B Marlowe wrote:
 Does anyone know if this feature is actually implemented? I just tried
 it with a Dial statement of mine and it doesn't play any file.  Doesn't
 report any errors, and I'm sure the file exists.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Matthew B
 Marlowe
 Sent: Tuesday, February 03, 2004 6:55 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Playing announcement to called user prior
 toConfirmation

 I wish 'A(x)' was available with AgentCallBackLogin!! :(

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Matteo
 Brancaleoni
 Sent: Tuesday, February 03, 2004 6:48 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Playing announcement to called user prior
 toConfirmation

 show application dial from asterisk cli:

 snip

   't' -- allow the called user transfer the calling user
   'T' -- to allow the calling user to transfer the call.
   'r' -- indicate ringing to the calling party, pass no audio until
 answered.
   'm' -- provide hold music to the calling party until answered.
   'H' -- allow caller to hang up by hitting *.
   'C' -- reset call detail record for this call.
   'P[(x)]' -- privacy mode, using 'x' as database if provided.
   'g' -- goes on in context if the destination channel hangs up
   'A(x)' -- play an announcement to the called party, using x as
 file

 see last param ...

 Matteo.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] busy tones

go with early b3

matteo.

Il mar, 2004-02-03 alle 12:44, Matteo Rancilio ha scritto:
 Hi
 
 When I call a phone with CAPI if the phone available I hear ringing ok 
 but if the phone is busy I don't hear anything at all.
 Also, when I call a mobile phone and it is turned off I don't hear the 
 operator voice answer me telling me that the request phone is turned off 
 or unavailable.
 
 Any ideas?
 
 m
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Dialling Hook Flash on Zaptel

2004-02-03 Thread Christopher Lee

Hi,

I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...

[FLASH] [*] [0] [22] (where 22 is the speed dial number)

But so far I've had no luck, with the following extension:-

exten = 922,1,Flash(${DIALOUTANALOG})
exten = 922,2,Dial(${DIALOUTANALOG}/*022)
exten = 922,3,Congestion
exten = 922,4,Hangup

Looking at the console, Asterisk doesn't get past the Flash command, telling
me that it's not a valid Zap channel. The call is being made from my Cisco
SIP phone through my local Asterisk Box, then via an IAX2 channel to the
site with the Asterisk box+X100P connected to the Norstar.

CONSOLE LOG
-- Accepting AUTHENTICATED call from 192.168.1.1, requested format = 2,
actual format = 2
-- Executing Flash([EMAIL PROTECTED]/2, Zap/1) in new stack Feb  3
22:37:19 WARNING[1146896]: app_flash.c:85 flash_exec:
[EMAIL PROTECTED]/2 is not a Zap channel
== Spawn extension (local, 922, 1) exited non-zero on
'[EMAIL PROTECTED]/2'
-- Hungup '[EMAIL PROTECTED]/2'
CONSOLE LOG

Is there some other way to dial a flash with the dial command? I notice
there's a W to insert a wait sequence.

Thanks,
Chris Lee


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] busy tones

2004-02-03 Thread Matteo Rancilio

Ciao Matteo,

I tried with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+CAPI+readmediff=2
My extensions are

[outgoing]
exten = 0,1,Goto(outgoing-isdn,s,1)
[outgoing-isdn]
exten = s,1,NoOp()
exten = _X.,1,Dial(CAPI/mynumber:b${EXTEN}|30)




Matteo Brancaleoni ha scritto:

go with early b3

matteo.

Il mar, 2004-02-03 alle 12:44, Matteo Rancilio ha scritto:
 

Hi

When I call a phone with CAPI if the phone available I hear ringing ok 
but if the phone is busy I don't hear anything at all.
Also, when I call a mobile phone and it is turned off I don't hear the 
operator voice answer me telling me that the request phone is turned off 
or unavailable.

Any ideas?

m

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
   



--
Matteo Rancilio
===
COMVERT S.R.L.
C.P. 211 - 20099 Sesto S. G. Centro (MI) - ITALY
Tel +39.02.27006796 | Fax +39.02.26005513
[EMAIL PROTECTED]
http://www.comvert.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Cisco 7940 SIP Registrations

2004-02-03 Thread Keith Lard

I am new to the list and I apologize for being late to the party.  I have a
couple of Cisco phones that I cannot get to register with *, any advise
would be appreciated.

Thanks

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Mediatrix 1102 Auth

Hi all.

I'm evaluating  a mediatrix 2fxs 1102.
seems great (it has also supervised transfer, that's
very needed in office environments and works well).
the only I thing I cannot make work is the auth
to my asterisk server.
If I don't set a password into the mediatrix and
*, I can call out, but still the registration goes wrong.
using a password, nothing works.

I've done some trace with ethereal, comparing the registration
process of one sip phone and of the mediatrix.
A sip phone registration normally works this way:
* phone tries to register
* asterisk sends out trying and then a proxy auth required
* the phone answers back with the logon data.
now the phone is registered.

the mediatrix stops at step2: never
answers to asterisk with logon data after a proxy auth required

any hint?

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Choppy Problem!!

Hi

 i'm managing a call center with asterisk, GS 102 and diva server 4 bri.
 
 i have big problem with big choppy sound, In the direction External
 user --- Agent

after a quick phone call with Cristian, we
managed to find out 2 things :
* hypertreading was enabled and that caused irq errors 
* capi.conf was wrong

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Details on TE410P Digium cards

Dan Iordanescu wrote:
[snip]
 1. How do you switch the card from ISDN PRI TE to NT? This means from
 being configured as User Equipment (the PABX) to be Network Equipment
 (the Exchange).

This is done in the /etc/asterisk/zapata.conf file.


 2. How do you configure the card for E1 or T1?

This is done with jumpers on the card, per port.

 3. In case of ISDN PRI over T1 Does it work with Line Type: PRI,
 channelised (23B + 1D) and CCS signalling? Where do you specify this
 setting?


Yes, in /etc/zaptel.conf

Mine looks like this:
;Incoming PRI
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

;Incoming PRI1
span=2,2,0,esf,b8zs
bchan=25-47
dchan=48
..

 4. Does it work with Trunk Type: TIE (incoming and outgoing calls)?
 Where do you set it?


dunno.  Never tried that.

 5. Where do you specify: Channel Type: voice-only


dunno what exactly you mean.  Mine has voice-only... and I didn't set
anything, so if that's what you mean...

 Thank you very much,
 Dan.

Hope this helps,
David Gomillion

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Cisco 7940 SIP Registrations

2004-02-03 Thread Andreas Hein

Hi,

that's my sip.conf entries for my Cisco 7060 Phones:

[general]
port = 5060 ; Port to bind to
bindaddr = my IP of * Server  ; Address to bind to
context = intern; Default for incoming calls
disallow=all; Disallow all codecs
allow=alaw  ; Allow codecs in order of preference
allow=ulaw  ; Allow codecs in order of preference
allow=ilbc
allow=gsm


[7000]
secret=secret
type=friend
context=intern
host=dynamic
mailbox=number
username=7000

and with SIP 6.0 Software on my Cisco Phones ...

it't works fine :-)
--
Andreas Hein


-- Original Message ---
From: Keith Lard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tue, 3 Feb 2004 08:41:58 -0500
Subject: [Asterisk-Users] Cisco 7940 SIP Registrations

 I am new to the list and I apologize for being late to the party.  I 
 have a couple of Cisco phones that I cannot get to register with *,
  any advise would be appreciated.
 
 Thanks
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--- End of Original Message ---

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Mediatrix 1102 Auth

 I'm evaluating  a mediatrix 2fxs 1102.
 seems great (it has also supervised transfer, that's
 very needed in office environments and works well).
 the only I thing I cannot make work is the auth
 to my asterisk server.
 If I don't set a password into the mediatrix and
 *, I can call out, but still the registration goes wrong.
 using a password, nothing works.
 
 I've done some trace with ethereal, comparing the registration
 process of one sip phone and of the mediatrix.
 A sip phone registration normally works this way:
 * phone tries to register
 * asterisk sends out trying and then a proxy auth required
 * the phone answers back with the logon data.
 now the phone is registered.
 
 the mediatrix stops at step2: never
 answers to asterisk with logon data after a proxy auth required
 
 any hint?

I've never played with the 1104, however others have reported that it
does register correctly when properly configured (and with * properly
matching).

In order for anyone to offer any suggestions, however, you'll have to
pass along the config info for both * and the 1104. Would suggest the
sip.conf entry (section) for one extension, and the relavent associated
entries for that extension programmed in the 1104. (no passwords please)



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

2004-02-03 Thread John Todd

At 10:59 PM +1000 2/3/04, Christopher Lee wrote:
Hi,

I'm trying to get my X100P to Dial the following sequence to gain access to
speed dial numbers on my Norstar PBX that the X100 is connected to...
[FLASH] [*] [0] [22] (where 22 is the speed dial number)

But so far I've had no luck, with the following extension:-

exten = 922,1,Flash(${DIALOUTANALOG})
exten = 922,2,Dial(${DIALOUTANALOG}/*022)
exten = 922,3,Congestion
exten = 922,4,Hangup
Looking at the console, Asterisk doesn't get past the Flash command, telling
me that it's not a valid Zap channel. The call is being made from my Cisco
SIP phone through my local Asterisk Box, then via an IAX2 channel to the
site with the Asterisk box+X100P connected to the Norstar.
CONSOLE LOG
-- Accepting AUTHENTICATED call from 192.168.1.1, requested format = 2,
actual format = 2
-- Executing Flash([EMAIL PROTECTED]/2, Zap/1) in new stack Feb  3
22:37:19 WARNING[1146896]: app_flash.c:85 flash_exec:
[EMAIL PROTECTED]/2 is not a Zap channel
== Spawn extension (local, 922, 1) exited non-zero on
'[EMAIL PROTECTED]/2'
-- Hungup '[EMAIL PROTECTED]/2'
CONSOLE LOG
Is there some other way to dial a flash with the dial command? I notice
there's a W to insert a wait sequence.
Thanks,
Chris Lee


Just for fun, try this:

exten = 922,1,Flash(Zap/1)
exten = 922,2,Dial(Zap/1/*022)
exten = 922,3,Congestion
exten = 922,4,Hangup
and see if it gives the same error.  I'd be interested to see if 
there's perhaps some strange variable swapping going on.

JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] dialing delay question.

John Bittner wrote:
 Hello.

 I have been working on getting my asterisk box to connect to a lucent
 definity PBX using a T100p. I connected it to a t1 port on the lucent

Let me start by saying I have not worked on a lucent definity.  Having said
that, I'll tell you my thoughts, and maybe they're things you have not yet
considered.

 call goes through. On the asterisk console I see the partial phone
 numbers when this happens.

It looks like the lucent is set to an immediate-type mode.  This is not very
surprising, as that is how my Nortel MICS acts.  It shouldn't be a really
big deal.


 How does this type of setup work. Is this a lucent issue or an
 asterisk issue. Does the lucent pbx just open a channel, send digits

I think that's what is happening.  I would try increasing the timeout in
your context that receives calls from the lucent.  I'm not sure right
off-hand how to do this, but when you find the command, I'd appreciate it,
as I'll probably have to fight the same battle in a couple of days.

Having said that, I'm looking at my Nortel, and can set how long the time
outs are.  Maybe you can do something like that for the Lucent?

I'd also take a good, long look at my extensions.conf file.  How does your
outgoing look?  Can Asterisk attempt to dial a phone number that is not
appropriately long?  i.e. Do you have an extension _9. or _X. or something
like that?  If so, try being more specific, like _1XX, or better
yet, use N's where you can.  This might keep Asterisk from attempting to
dial before it's time.

Final question: does the Lucent T1 card also support PRI?  That's what we're
using between our Nortel and *, and so far it seems to work well.

 Any help will be appreciated.


Hope this helps.  And please, hold the flames to a minimum.  I told you I
didn't know for sure what to try, but thought these might get you on the
right path.

Thanks,
David Gomillion

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors

2004-02-03 Thread Geert Nijpels

Hi All,

I have been busy with this problem for a while now, but I can't find any 
solution. First I thought this was a problem with the phones, but all my 
phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried 
all firmware versions I could find for the phones.

First, my situation:
- No NAT, No Firewall, same subnet
- Codec configuration:
In general:
disallow=all
disallow=g723.1
disallow=g729
disallow=gsm
allow=ulaw
allow=alaw
In the phones:
disallow=all
disallow=g723.1
disallow=g729
disallow=gsm
allow=ulaw
allow=alaw
But I also tried other codec configs. (allow=gsm, etc). Same problem. 
I'm testing from the Cisco 7960, as this phone seems to work best. I 
could also test from another phone with the same results. The S is for 
Success (can talk), the F is for Failure(Call gets setup but no 
speech/sound).

Cisco 7960 to SNOM
S,S,F,S,F,F,F,S,S,S,S,S,F
Cisco 7960 to GS
S,F,S,S,F,S,S,F,S,F,
I placed a sip debug from asterisk for each situation at the following URL:

http://audix.noc.ams-ix.net/asterisk/dumps/

- cisco_to_gs_failure.txt
- cisco_to_gs_success.txt
- cisco_to_snom_success.txt
- cisco_to_snom_failure.txt
Somebody have a clue? I'm thinking of filing a bug but I want to make 
sure this is no configuration or other problem at my side.

Thanks and kind regards,

Geert Nijpels

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors

2004-02-03 Thread Eric Wieling

Asterisk is still saying it accepts G729.  That is prolly the problem. 
Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1.

If there any reason you are allowing both ulaw AND alaw.

On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote:
 Hi All,
 
 I have been busy with this problem for a while now, but I can't find any 
 solution. First I thought this was a problem with the phones, but all my 
 phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried 
 all firmware versions I could find for the phones.
 
 First, my situation:
 - No NAT, No Firewall, same subnet
 - Codec configuration:
 
 In general:
 disallow=all
 disallow=g723.1
 disallow=g729
 disallow=gsm
 allow=ulaw
 allow=alaw
 
 In the phones:
 disallow=all
 disallow=g723.1
 disallow=g729
 disallow=gsm
 allow=ulaw
 allow=alaw
 
 But I also tried other codec configs. (allow=gsm, etc). Same problem. 
 I'm testing from the Cisco 7960, as this phone seems to work best. I 
 could also test from another phone with the same results. The S is for 
 Success (can talk), the F is for Failure(Call gets setup but no 
 speech/sound).
 
 Cisco 7960 to SNOM
 S,S,F,S,F,F,F,S,S,S,S,S,F
 
 Cisco 7960 to GS
 S,F,S,S,F,S,S,F,S,F,
 
 I placed a sip debug from asterisk for each situation at the following URL:
 
 http://audix.noc.ams-ix.net/asterisk/dumps/
 
 - cisco_to_gs_failure.txt
 - cisco_to_gs_success.txt
 - cisco_to_snom_success.txt
 - cisco_to_snom_failure.txt
 
 Somebody have a clue? I'm thinking of filing a bug but I want to make 
 sure this is no configuration or other problem at my side.
 
 Thanks and kind regards,
 
 Geert Nijpels
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
Asterisk Resource Pages.

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

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Playing announcement to called user prior toConfirmation

I have a Dial Statement and at the end ,m,A(transfer) but when the
extension picks up it doesn't play anything

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of reseaux
Sent: Tuesday, February 03, 2004 7:14 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Playing announcement to called user prior
toConfirmation

Dear Matthew
yes it work great A(playmex) where playmex is gsm file in sound
dir.. i have 
made some simple hack to app_dial.c to have a new option B(playmex) with
it i 
can play a mex to the caller when the call is connected i use it to play
a 
dtmf code...
Thanks in advance
Dimitri

PS: if someone think is good option a send it

On Tuesday 03 February 2004 12:21, Matthew B Marlowe wrote:
 Does anyone know if this feature is actually implemented? I just tried
 it with a Dial statement of mine and it doesn't play any file.
Doesn't
 report any errors, and I'm sure the file exists.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Matthew B
 Marlowe
 Sent: Tuesday, February 03, 2004 6:55 AM
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Playing announcement to called user
prior
 toConfirmation

 I wish 'A(x)' was available with AgentCallBackLogin!! :(

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Matteo
 Brancaleoni
 Sent: Tuesday, February 03, 2004 6:48 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Playing announcement to called user
prior
 toConfirmation

 show application dial from asterisk cli:

 snip

   't' -- allow the called user transfer the calling user
   'T' -- to allow the calling user to transfer the call.
   'r' -- indicate ringing to the calling party, pass no audio
until
 answered.
   'm' -- provide hold music to the calling party until answered.
   'H' -- allow caller to hang up by hitting *.
   'C' -- reset call detail record for this call.
   'P[(x)]' -- privacy mode, using 'x' as database if provided.
   'g' -- goes on in context if the destination channel hangs up
   'A(x)' -- play an announcement to the called party, using x as
 file

 see last param ...

 Matteo.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Still looking for small fxo sip gateway


I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank  T1 card suggestions, 
please. I've also just completed an eval of the Mediatrix 1204 which
does not support the requirements.)

The market between two fxo pstn lines (pair of x100p's) and something
around four to six lines seems to be lacking, or I'm looking in the
wrong search engine (or something). I fully understand the economics of
when a channel bank and T1 card becomes cost effective, including the 
eBay costs (and risks), etc. I've also heard the comments for months 
now that Digium is/will be selling something real-soon-now.

Specifically, I'd like to use a 4-port fxo sip gateway capable of supporting
four US pstn analog lines, CallerID, Touchtone, loop style supervision,
and have the capability for asterisk to direct an outbound call to a 
specific port on that gateway. I think that implies each port must
execute a sip register command successfully. It's also expected to accept 
incoming pstn calls directing those to a single asterisk. (I don't care 
about an IP dialtone, nat, etc, just a plain-jane two-way sip gateway.)

If anyone is designing such a box and need professional eval, we can 
certainly work with you privately (off list to radamson @ routers dot com)
to accomidate those needs.

Anyone seen such a beast at a reasonable price?

Rich



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Problems with chan_sip: random calls have no sound withouth any errors

2004-02-03 Thread Geert Nijpels

Eric Wieling wrote:

Asterisk is still saying it accepts G729.  That is prolly the problem. 
Try updating to the latest CVS of Asterisk or to Asterisk version 0.7.1.

If there any reason you are allowing both ulaw AND alaw.
 

Sorry forgot to mention it. I'm already at latest CVS, but I have this 
problem also with 0.7.1. Well I use alaw and ulaw because all my phones 
support these codecs. But I get this problem with other codec 
configurations too.

Kind regards,

Geert

On Tue, 2004-02-03 at 08:48, Geert Nijpels wrote:
 

Hi All,

I have been busy with this problem for a while now, but I can't find any 
solution. First I thought this was a problem with the phones, but all my 
phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried 
all firmware versions I could find for the phones.

First, my situation:
- No NAT, No Firewall, same subnet
- Codec configuration:
In general:
disallow=all
disallow=g723.1
disallow=g729
disallow=gsm
allow=ulaw
allow=alaw
In the phones:
disallow=all
disallow=g723.1
disallow=g729
disallow=gsm
allow=ulaw
allow=alaw
But I also tried other codec configs. (allow=gsm, etc). Same problem. 
I'm testing from the Cisco 7960, as this phone seems to work best. I 
could also test from another phone with the same results. The S is for 
Success (can talk), the F is for Failure(Call gets setup but no 
speech/sound).

Cisco 7960 to SNOM
S,S,F,S,F,F,F,S,S,S,S,S,F
Cisco 7960 to GS
S,F,S,S,F,S,S,F,S,F,
I placed a sip debug from asterisk for each situation at the following URL:

http://audix.noc.ams-ix.net/asterisk/dumps/

- cisco_to_gs_failure.txt
- cisco_to_gs_success.txt
- cisco_to_snom_success.txt
- cisco_to_snom_failure.txt
Somebody have a clue? I'm thinking of filing a bug but I want to make 
sure this is no configuration or other problem at my side.

Thanks and kind regards,

Geert Nijpels

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
   

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Using a Dial Statement with option m and t









When I use option t and m together in the same dial
statement the music on hold doesnt appear to work.



Is this a normal operation?

[Asterisk-Users] SIP debug logs

2004-02-03 Thread Steve Foy

This strikes me as something that should be really very simple to do, but I
can't figure it out.

Is there a way of logging all SIP debuging info to a file somewhere?

It would help me greatly!

Cheers,
Steve

-- 
Steve Foy|  http://www.unite.net
UNITE Solutions  |  Tel: 028 9077 7338 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
 Hi,
 
 I'm trying to get my X100P to Dial the following sequence to gain access to
 speed dial numbers on my Norstar PBX that the X100 is connected to...
 
 [FLASH] [*] [0] [22] (where 22 is the speed dial number)
 
 But so far I've had no luck, with the following extension:-
 
 exten = 922,1,Flash(${DIALOUTANALOG})
 exten = 922,2,Dial(${DIALOUTANALOG}/*022)
 exten = 922,3,Congestion
 exten = 922,4,Hangup
 
 Looking at the console, Asterisk doesn't get past the Flash command, telling
 me that it's not a valid Zap channel. The call is being made from my Cisco
 SIP phone through my local Asterisk Box, then via an IAX2 channel to the
 site with the Asterisk box+X100P connected to the Norstar.
 
 CONSOLE LOG
 -- Accepting AUTHENTICATED call from 192.168.1.1, requested format = 2,
 actual format = 2
 -- Executing Flash([EMAIL PROTECTED]/2, Zap/1) in new stack Feb  3
 22:37:19 WARNING[1146896]: app_flash.c:85 flash_exec:
 [EMAIL PROTECTED]/2 is not a Zap channel
 == Spawn extension (local, 922, 1) exited non-zero on
 '[EMAIL PROTECTED]/2'
 -- Hungup '[EMAIL PROTECTED]/2'
 CONSOLE LOG
 
 Is there some other way to dial a flash with the dial command? I notice
 there's a W to insert a wait sequence.


The problem with your example is that a flash must be executed after you
have a channel since otherwise there is no offhook event to then be
toggled by the on then off hook, you just would be off hook. Second,
look at the documentation for flash and you will see that the flash
command doesn't accept a argument.

As to how to actually accomplish what you want, I don't know how.  
-- 
Steven Critchfield [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP debug logs

2004-02-03 Thread Geert Nijpels

Steve Foy wrote:

This strikes me as something that should be really very simple to do, but I
can't figure it out.
Is there a way of logging all SIP debuging info to a file somewhere?

It would help me greatly!
 

I dont know if it's possible using asterisk. You can use the command 
'script -a filename' that will record everything at the prompt.

Geert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP debug logs

2004-02-03 Thread Regovich, Timothy

Or you could modify the logger and have all SIP messages set at a different
log level and have them go to a file (/var/log/messages/sip) for example.

Tim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Geert Nijpels
Sent: Tuesday, February 03, 2004 11:38 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP debug logs


Steve Foy wrote:

This strikes me as something that should be really very simple to do, but I
can't figure it out.

Is there a way of logging all SIP debuging info to a file somewhere?

It would help me greatly!
  

I dont know if it's possible using asterisk. You can use the command 
'script -a filename' that will record everything at the prompt.

Geert
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



--
Notice:  This e-mail message, together with any attachments, contains information of 
Merck  Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or 
its affiliates (which may be known outside the United States as Merck Frosst, Merck 
Sharp  Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary 
copyrighted and/or legally privileged. It is intended solely for the use of the 
individual or entity named on this message.  If you are not the intended recipient, 
and have received this message in error, please notify us immediately by reply e-mail 
and then delete it from your system.
--
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] kernel 2.4.x .... which one?

2004-02-03 Thread mihai iancu

Hello,

I use 2.4.18-14 and as soon as I did CVS after Jan 10th, 2004
everything went wrong in terms of compiling zaptel.

No matter what I get compiling errors related to different header files
from linux kernel source tree.

Which kernel version you guys used when you tested the latest zaptel
available for us via CVS?

Thank you,

Mihai Iancu

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Using a Dial Statement with option m and t

2004-02-03 Thread John Todd

When I use option t and m together in the same dial statement the 
music on hold doesn't appear to work.



Is this a normal operation?


1) Please don't post with HTML.  Read the archives for several 
lengthy flamewars over this topic.  Comments as to how I suck because 
I don't like HTML will be ignored (this comment not directed at you, 
Matthew.)

2) Please learn to ask complete questions:

  -Include all relevant lines of your extensions.conf file.
  -Include a small clip of the console output that surrounds the 
event where you have difficulty

I suspect I know what lies at the root of your problem, but I don't 
want to spend my time guessing.  We're happy to help here, but you 
need to give more data so we can point out where you are doing 
something wrong, or if you're not doing something wrong, we need to 
see everything so we can determine if this is a bug.

JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Mediatrix 1102 Auth

Hi.
  any hint?
 
 I've never played with the 1104, however others have reported that it
 does register correctly when properly configured (and with * properly
 matching).
 
 In order for anyone to offer any suggestions, however, you'll have to
 pass along the config info for both * and the 1104. Would suggest the
 sip.conf entry (section) for one extension, and the relavent associated
 entries for that extension programmed in the 1104. (no passwords please)

I managed to make it work.
I simply wrote the wrong real into the meadiatrix, since I wrote
the * ip addr, instead of asterisk.
reverting that, made it register without issues.

Matteo.

-- 
Matteo Brancaleoni
Espia System Administrator
Email : [EMAIL PROTECTED]
Web   : http://www.espia.it
Phone : +39 02 70633354  - ext 201
IAX(2): [EMAIL PROTECTED] - ext 201
Iaxtel: 1-700-56-62458   - ext 201

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] upgrade problems

2004-02-03 Thread Chris Lee

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was 
relesed.
now I am having troubles with my dialing plan and voice mail.

As part of the upgrade I re-built the machine so there was a blank slate 
however after installing 0.7.1 I had no mail box creation script and 
could not figure out how to go about creating a mailbox, any suggestions 
would be usefull.
I have looked at tall the docs I can find and the closest I get tells me 
to use the script to create a mailbox, but there is no script.

The second problem I am having is that my dialplan is not working how it 
used to, for reasons I cant quite put my fingure on;
here is the dial plan:

[ Context 'default' created by 'pbx_config' ]
 's' =1. Goto(well|3000|1)   [pbx_config]




[ Context 'in-sip' created by 'pbx_config' ]
 Include ='voicemail access'
[pbx_config]


[ Context 'users' created by 'pbx_config' ]
 '5000' = 1. Hangup()   
[pbx_config]
 '5001' = 1. Hangup()   
[pbx_config]




[ Context 'well' created by 'pbx_config' ]
 'h' =1. Hangup()   
[pbx_config]


 Include ='emergency'   
[pbx_config]
 Include ='voicmail access' 
[pbx_config]
 Include ='external access' 
[pbx_config]
 Include ='extensions'  
[pbx_config]
 Include ='no match'
[pbx_config]
  

[ Context 'extensions' created by 'pbx_config' ]
 '3000' = 1. Dial(${p1}|10|tr)  
[pbx_config]
   2. Answer()   
[pbx_config]
   3. Background(vm/3000/unavail)
[pbx_config]
   4. Voicemail(3000)
[pbx_config]
   5. Hangup()   
[pbx_config]
   102. Background(vm/3000/unavail)  
[pbx_config]
   103. Goto(4)  
[pbx_config]
 '3001' = 1. Macro(Standard-Ext|${p2})  
[pbx_config]
 '3002' = 1. Macro(Standard-Ext|${p3})  
[pbx_config]




[ Context 'no match' created by 'pbx_config' ]
 '_.' =   1. Playback(sorry-no-match)   
[pbx_config]
   2. Hangup()   
[pbx_config]




[ Context 'external access' created by 'pbx_config' ]
 '_9.' =  1. Dial(${Line1}/$(EXTEN:1))  
[pbx_config]


 Ignore pattern = '9'   
[pbx_config]


[ Context 'emergency' created by 'pbx_config' ]
 '112' =  1. Dial(${line1}/${emergency})
[pbx_config]
   102. Hangup(${line1}) 
[pbx_config]
   103. Goto(1)  
[pbx_config]
 '999' =  1. Dial(${line1}/${emergency})
[pbx_config]
   102. Hangup(${line1})

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

Steven Critchfield wrote:
 On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
 Hi,

 I'm trying to get my X100P to Dial the following sequence to gain
 access to speed dial numbers on my Norstar PBX that the X100 is
 connected to...
[snip]
 The problem with your example is that a flash must be executed after
 you have a channel since otherwise there is no offhook event to then
 be toggled by the on then off hook, you just would be off hook.
 Second, look at the documentation for flash and you will see that the
 flash command doesn't accept a argument.

 As to how to actually accomplish what you want, I don't know how.

since you're connecting an X100P to a Norstar PBX, maybe immediate mode
would work?  I'm not sure how well that would work with a SIP client like
you were talking about...

I had some similar problems with the X100P and our ATA-2.  I also couldn't
ever get the Nortel to recognize the DTMF, or get Asterisk to recognize DTMF
coming through the Nortel.  I wish I could say that I figured out a really
cool way to make it work, but instead I moved on and interconnected via
PRIs.

How often do your speed dials change?  If it's not very often, maybe you
should recreate it in *.  You could have an extension which sends you into a
context that asks for the speed dial code.  You could then key it in, and it
would send you where you want to go.  Not elegant, but it might be good
enough.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cdr mysql problem

On Tuesday 03 February 2004 01:44, Tomica Crnek wrote:
 [global]
 hostname=localhost
 dbname=asteriskcdrdb
 password=**
 user=asteriskcdruser
 ;port=3306
 ;sock=/tmp/mysql.sock
 sock=/var/lib/mysql/mysql.sock

Okay, and so does this work?

bash$ echo select max(calldate) from cdr; | \
 mysql -uasteriskcdruser -S/var/lib/mysql/mysql.sock \
 -p asteriskcdrdb
Enter password:
max(calldate)
2004-02-02 19:19:22
bash$

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Still looking for small fxo sip gateway

2004-02-03 Thread Kostur, Andre

Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway

You might want to take a look on the Wiki pages for VoIP, in particular:

http://www.voip-info.org/wiki-VoIP+Gateways

Offhand at our site we're trying to set up something similar (although a little larger, 10 FXO lines, but no requirement to pick which line the call goes out... our 10 lines are all overlines). Our Vegastream 50 FXO shipped yesterday (or perhaps this morning), so we should be getting it in a day or two. (BTW: I'm in Canada)

There's been rumours posted to this list that Digium is coming out with a higher-density FXO card, and Woody mentioned a Voicetronix Openline12, which appears to be a 12-port FXO card. And I believe that Intel/Dialogic puts out some multiport FXS/FXO cards...

-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, February 03, 2004 6:15 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Still looking for small fxo sip gateway

I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up
with anything
that fits. Suggestions anyone? (No channel bank T1 card
suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
does not support the requirements.)

The market between two fxo pstn lines (pair of x100p's) and something
around four to six lines seems to be lacking, or I'm looking in the
wrong search engine (or something). I fully understand the
economics of
when a channel bank and T1 card becomes cost effective, including the
eBay costs (and risks), etc. I've also heard the comments for months
now that Digium is/will be selling something real-soon-now.

Specifically, I'd like to use a 4-port fxo sip gateway
capable of supporting
four US pstn analog lines, CallerID, Touchtone, loop style
supervision,
and have the capability for asterisk to direct an outbound call to a
specific port on that gateway. I think that implies each port must
execute a sip register command successfully. It's also
expected to accept
incoming pstn calls directing those to a single asterisk. (I
don't care
about an IP dialtone, nat, etc, just a plain-jane two-way sip
gateway.)

If anyone is designing such a box and need professional eval, we can
certainly work with you privately (off list to radamson @
routers dot com)
to accomidate those needs.

Anyone seen such a beast at a reasonable price?

[Asterisk-Users] Asterisk 0.7.1 RPMS Updated to Rel 4

2004-02-03 Thread Greg Boehnlein

Neo: What are you trying to tell me? That I can dodge bullets?

Morpheus: No, Neo. I'm trying to tell you that when you're ready,
you won't have to.

There have been over 500 downloads of the RedHat Asterisk RPMS 
since they were released 2 weeks ago, and I have received many comments 
to improve them. After some late night hacking this weekend, I have 
dropped 0.7.1 release 4 RPMS at ftp://ftp.nacs.net/asterisk.
This is the first release that I feel is usable by the general 
public. Having got my hands on some Digium hardware, I was able to see 
that my build environments for RH73,9 and FC1 were generating i686 
specific modules for Zaptel, which made the RPMS unusable on standard i386 
kernels. Since most people never bother to put the i686 kernel, I 
downgraded my build environments and rebuilt the packages. I.E. if you are 
using stock RedHat kernels (kernel*.i386.rpm) then these are build for 
you. If not, you'll have to grab the .src.rpm and rebuild the RPMS for 
yourself.
Also, based on feedback from Donnie Barnes and Brian West, I 
added the result of make config from the Zaptel RPMS which will load the 
correct modules at startup.
As always, comments and suggestions are welcomed and appreciated. 
I could use some volunteers to build/test i586, i686 and k7 packages 
or RH9 and FC1, so if you are interested drop me a line.

Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM

Current Release
---
asterisk-0.7.1-4.i386.rpm
libpri-0.5.1-4.i386.rpm
zaptel-0.8.0-4.i386.rpm
kernel-module-zaptel-0.8.0-4_2.4.20_28.7.i386.rpm

FTP Download

ftp://ftp.nacs.net/asterisk/

Changelog
-
* Sat Jan 31 2004 Gregory Boehnlein [EMAIL PROTECTED]

- Updated development environment to ensure proper build consistency for chan_zap
- Added post-install chkconfig to auto-start asterisk on boot
- First really useable release. Yay!

* Mon Jan 26 2004 Gregory Boehnlein [EMAIL PROTECTED]

- Updated changelog entry to enable build on Fedora Core 1 [EMAIL PROTECTED]
- Made the decsision to use Dist Specific version numbers (_fc1,_rh9,_rh8,_rh73)

* Sat Jan 24 2004 Gregory Boehnlein [EMAIL PROTECTED]

- added doc macros
- added config macros
- updated install stanza to correct symlink issue
- updated patch0 to include changes to Makefile
- added /etc/rc.d/init.d/asterisk
- added export LD_ASSUME_KERNEL=2.4.1 for RH9
- asterisk.spec now builds cleanly on RH73 and RH9

* Wed Jan 21 2004 Gregory J. Boehnlein [EMAIL PROTECTED] 

- Initial .spec file created. Most likely buggered. Badly needs help.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP debug logs

 This strikes me as something that should be really very simple to do, but I
 can't figure it out.
 
 Is there a way of logging all SIP debuging info to a file somewhere?
 
 It would help me greatly!

Might take a look at /etc/asterisk/logger.conf file to see if that's what
you're looking for. Seems to me I added the debug level some time ago to
diagnose a specific problem,  and the /var/log/asterisk/debug log file 
grew large very quickly and included a ton of detail.

; Logging Configuration  
;   
[logfiles] 
;  
; Format is filename and then levels of debugging to be included:
;debug
;notice
;warning 
;error  
;
; Special filename console represents the system console
;
; debug = debug
console = notice,warning,error  
messages = notice,warning,error   


Rich


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Cisco 7960 quick dial

2004-02-03 Thread Jose Inzunza/YM/RWDOE

Is there a way to make  the Cisco 7960 SIP phone dial out automatically
without having to press the dial button, once the numbers that you have
entered match a specific pattern?  This feature is present when the phone
is working with a Cisco CallManager.  For example, if all of my internal
extensions begin with a '5' and are four digits long, if I dialed '5123' on
the phone, the call would initiate once I pressed the '3'.  Any help would
be appreciated.

Jose


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Still looking for small fxo sip gateway

2004-02-03 Thread Kannaiyan Natesan

Title: RE: [Asterisk-Users] Still looking for small fxo sip gateway

The wonder is none of the FXO devices works fine except
asterisk X100P.

I'm not sure what is the stupidity present in that analog
technology.

Kannaiyan

- Original Message -
From:
Kostur,
Andre
To: '[EMAIL PROTECTED]'

Sent: Tuesday, February 03, 2004 3:58
PM
Subject: RE: [Asterisk-Users] Still
looking for small fxo sip gateway

You might want to take a look on the Wiki pages for VoIP, in
particular:
http://www.voip-info.org/wiki-VoIP+Gateways
Offhand at our site we're trying to set up something similar
(although a little larger, 10 FXO lines, but no requirement to pick which line
the call goes out... our 10 lines are all overlines). Our Vegastream
50 FXO shipped yesterday (or perhaps this morning), so we should be getting it
in a day or two. (BTW: I'm in Canada)
There's been rumours posted to this list that Digium is coming
out with a higher-density FXO card, and Woody mentioned a Voicetronix
Openline12, which appears to be a 12-port FXO card. And I believe that
Intel/Dialogic puts out some multiport FXS/FXO cards...
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]]
Sent: Tuesday, February 03, 2004 6:15 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Still looking for small fxo sip gateway
I've been looking around for a small
external sip fxo gateway, sending emails to
possible vendors, etc, and can not seem to come up with anything that fits. Suggestions
anyone? (No channel bank T1 card
suggestions, please. I've also just completed an
eval of the Mediatrix 1204 which does not support
the requirements.)
The market between two fxo pstn lines (pair of x100p's) and something
around four to six lines seems to be lacking, or I'm
looking in the wrong search engine (or
something). I fully understand the economics
of when a channel bank and T1 card becomes cost
effective, including the eBay costs (and risks),
etc. I've also heard the comments for months now
that Digium is/will be selling something real-soon-now. Specifically, I'd like to use a
4-port fxo sip gateway capable of
supporting four US pstn analog lines, CallerID,
Touchtone, loop style supervision,
and have the capability for asterisk to direct an
outbound call to a specific port on that gateway.
I "think" that implies "each" port must execute a
sip register command successfully. It's also
expected to accept incoming pstn calls directing
those to a single asterisk. (I don't care
about an IP dialtone, nat, etc, just a plain-jane
two-way sip gateway.) If anyone is designing such a box and
need professional eval, we can certainly work
with you privately (off list to radamson @
routers dot com) to accomidate those
needs. Anyone seen
such a beast at a reasonable price?

[Asterisk-Users] Nortel and Asterisk interconnection

I have created a pdf document about my experience in integrating a Nortel
Norstar MICS with *.  This is not a cookbook, but it does describe the
process I followed and gave a copy of the relevant configuration files.

If anybody is interested, please feel free to download a copy at
http://www.eyecarenow.com/asterisk.

Please be patient, as the Internet connection here is, well, lacking.  If
anybody finds this useful and would like to mirror it, please let me know.

Thanks,
David Gomillion

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Cisco 7960 quick dial

2004-02-03 Thread Bisker, Scott (7805)

Take a look at dialplan.xml on your tftp server.




DIALTEMPLATE
TEMPLATE MATCH=0  Timeout=1 User=IP/ !-- Local operator--
TEMPLATE MATCH=8,011* Timeout=6 User=IP/ !-- International 
calls--
TEMPLATE MATCH=8,1..  Timeout=0 User=IP/ !-- Long Distance --
TEMPLATE MATCH=9,1..  Timeout=0 User=IP/ !-- Toll Free --
TEMPLATE MATCH=9,11   Timeout=0 User=IP Route=Emergency 
Rewrite=9911/
TEMPLATE MATCH=9,..   Timeout=0 User=IP/ !-- Local numbers --
TEMPLATE MATCH=9,.11  Timeout=0 User=IP/ !-- Service numbers --
TEMPLATE MATCH=78..   Timeout=1 User=IP/ !-- Corporate Dial 
plan--
TEMPLATE MATCH=52..   Timeout=1 User=IP/ !-- Corporate Dial 
plan--
TEMPLATE MATCH=87..   Timeout=1 User=IP/ !-- Corporate Dial 
plan--
TEMPLATE MATCH=5000   Timeout=1 User=IP/ !-- Voicemail --
TEMPLATE MATCH=4...   Timeout=1 User=IP/ !--  Meetme --
TEMPLATE MATCH=11..   Timeout=1 User=IP/ !-- Parking --
TEMPLATE MATCH=*  Timeout=15/ !-- Anything else --
TEMPLATE MATCH=123#45#6   Timeout=0 User=IP/ !-- Match `#' --
TEMPLATE MATCH=12\*345Timeout=0 User=IP/ !-- Match * Char --
/DIALTEMPLATE



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jose
Inzunza/YM/RWDOE
Sent: Tuesday, February 03, 2004 11:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cisco 7960 quick dial


Is there a way to make  the Cisco 7960 SIP phone dial out automatically
without having to press the dial button, once the numbers that you have
entered match a specific pattern?  This feature is present when the phone
is working with a Cisco CallManager.  For example, if all of my internal
extensions begin with a '5' and are four digits long, if I dialed '5123' on
the phone, the call would initiate once I pressed the '3'.  Any help would
be appreciated.

Jose


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Still looking for small fxo sip gateway

2004-02-03 Thread John Todd

I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank  T1 card suggestions,
please. I've also just completed an eval of the Mediatrix 1204 which
does not support the requirements.)
The market between two fxo pstn lines (pair of x100p's) and something
around four to six lines seems to be lacking, or I'm looking in the
wrong search engine (or something). I fully understand the economics of
when a channel bank and T1 card becomes cost effective, including the
eBay costs (and risks), etc. I've also heard the comments for months
now that Digium is/will be selling something real-soon-now.
Specifically, I'd like to use a 4-port fxo sip gateway capable of supporting
four US pstn analog lines, CallerID, Touchtone, loop style supervision,
and have the capability for asterisk to direct an outbound call to a
specific port on that gateway. I think that implies each port must
execute a sip register command successfully. It's also expected to accept
incoming pstn calls directing those to a single asterisk. (I don't care
about an IP dialtone, nat, etc, just a plain-jane two-way sip gateway.)
If anyone is designing such a box and need professional eval, we can
certainly work with you privately (off list to radamson @ routers dot com)
to accomidate those needs.
Anyone seen such a beast at a reasonable price?

Rich
Rich -
  No.  :-)  However, you might consider Welltech ( 
http://www.welltech.com.tw/) to see if you can demo their products - 
they have a SIP load now for some of their boxes.  Website currently 
offline, but I think that's just a network error not a company 
error.

JT

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Greg Boehnlein

Hello all,
Saturday night, after a couple of shots of bourbon, I realized 
that I had an old PC sitting in the garage that I could use as an Asterisk 
gateway if I just blew the dust off it and reloaded it with a modern Linux 
distribution. In my characteristically impulsive manner, I grabbed it and 
started cleaning it up so that I could put it in my office without my wife 
having a fit.
The sytem is an old Gateway system, that I used to use as an 
X-terminal. Nothing special really, P-133, 16 megs of ram, 3 PCI slots, 
3.2 gig hard drive. The box booted and I was greated with a RH9 login 
screen from my X-server.
After imaging the hard drive over to my server for backup 
purposes, I proceeded to try installing Fedora, RH9, RH8 and finally RH73 
without any luck. The 16 megs of ram was just too small to do the 
installation. So I grabbed a Debian 3.0 netinstall image and got the box 
online and running.
8 hours later, apt-get dist-upgrade completed and the box was 
running Debian 3.0 unstable. Now it was time install Asterisk. An 
apt-cache search asterisk revealed that Debian unstable has pkg files 
available. Yay! That'll save me the time of bulding everything on this 
box so all I will need to do is rebuild the Zaptel modules.
20 minutes later, I had my Zaptel modules built and was ready to 
give it a whirl, so I loaded the wcfxo module and started Asterisk. My 
GrandStream registered against the server and I was able to able to place 
calls out the PSTN using the box.
Initially, I was prepared for this to be an excercise in futility, 
but I have been extremely surprised by the results. I can support up to 3 
concurrent SIP sessions before I start to get degraded quality, and the 
box appears to be rock solid. I have it registered against our production 
Asterisk server at work over my Cable modem, and my staff can simply dial 
3xxx to ring my extension at home. Voicemail works just fine and with the 
addition of the Asterisk-sounds pkg inbond callers now know that we are 
out Gambling and getting drunk when they call.

Is this the smallest Asterisk server ever? :)

asterisk:~# cat /proc/cpuinfo 
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 5
model   : 2
model name  : Pentium 75 - 200
stepping: 12
cpu MHz : 132.957
fdiv_bug: no
hlt_bug : no
f00f_bug: yes
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 1
wp  : yes
flags   : fpu vme de pse tsc msr mce cx8
bogomips: 265.42

asterisk:~# free
 total   used   free sharedbuffers cached
Mem: 13984  13696288  0   1372868
-/+ buffers/cache:  11456   2528
Swap:92728  17316  75412

asterisk:~# ps aux
USER   PID %CPU %MEM   VSZ  RSS TTY  STAT START   TIME COMMAND
root 1  0.0  0.6  1492   84 ?SFeb02   0:00 init [2]   
root 2  0.0  0.0 00 ?SW   Feb02   0:00 [keventd]
root 3  0.0  0.0 00 ?SWN  Feb02   0:00 [ksoftirqd_CPU0]
root 4  0.0  0.0 00 ?SW   Feb02   0:14 [kswapd]
root 5  0.0  0.0 00 ?SW   Feb02   0:00 [bdflush]
root 6  0.0  0.0 00 ?SW   Feb02   0:00 [kupdated]
root85  0.0  0.0 00 ?DW   Feb02   0:01 [kjournald]
root   292  0.0  1.1  1540  164 ?SFeb02   0:00 /sbin/syslogd
root   295  0.0  0.0  21564 ?SFeb02   0:01 /sbin/klogd
root   309  0.0  0.0  15200 ?SW   Feb02   0:00 /usr/sbin/inetd
root   316  0.0  0.4  3064   56 ?SFeb02   0:00 /usr/sbin/sshd
root   325  0.0  0.9  1752  128 ?SFeb02   0:00 /usr/sbin/cron
root   329  0.0  0.4  1488   56 tty1 SFeb02   0:00 /sbin/getty 38400 tty1
root   330  0.0  0.4  1488   56 tty2 SFeb02   0:00 /sbin/getty 38400 tty2
root  2609  0.0  0.2  2276   40 ?SFeb02   0:00 /bin/sh 
/usr/sbin/safe_asterisk
root  2611  0.0  7.3 42144 1032 ?SFeb02   0:03 asterisk -vvvg -c
root  2612  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2613  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2614  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2615  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2616  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2617  0.0  7.3 42144 1032 ?SFeb02   0:21 asterisk -vvvg -c
root  2618  0.0  7.3 42144 1032 ?SFeb02   0:13 asterisk -vvvg -c
root  2619  0.0  7.3 42144 1032 ?SFeb02   0:00 asterisk -vvvg -c
root  2620  0.4  7.3 42144 1032 ?SFeb02   7:52 asterisk -vvvg -c
root  2621  0.0  7.3 42144 1032 ?SFeb02

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
 Steven Critchfield wrote:
  On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
  Hi,
 
  I'm trying to get my X100P to Dial the following sequence to gain
  access to speed dial numbers on my Norstar PBX that the X100 is
  connected to...
 [snip]
  The problem with your example is that a flash must be executed after
  you have a channel since otherwise there is no offhook event to then
  be toggled by the on then off hook, you just would be off hook.
  Second, look at the documentation for flash and you will see that the
  flash command doesn't accept a argument.
 
  As to how to actually accomplish what you want, I don't know how.
 
 since you're connecting an X100P to a Norstar PBX, maybe immediate mode
 would work?  I'm not sure how well that would work with a SIP client like
 you were talking about...

Not to flame, you but you really need to work on those critical reading
skills. The example message that was quoted till you trimmed it had IAX2
as the VoIP not SIP. 

Next, Immediate mode only is of use when you have dialed, or been
dialed. We are still in the state of not having dialed and therefore
selected a outbound line. 

 How often do your speed dials change?  If it's not very often, maybe you
 should recreate it in *.  You could have an extension which sends you into a
 context that asks for the speed dial code.  You could then key it in, and it
 would send you where you want to go.  Not elegant, but it might be good
 enough.

This is probably the only useful suggestion, and most likely the only
one to work.

-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Cisco 7960 quick dial

 Is there a way to make  the Cisco 7960 SIP phone dial out automatically
 without having to press the dial button, once the numbers that you have
 entered match a specific pattern?  This feature is present when the phone
 is working with a Cisco CallManager.  For example, if all of my internal
 extensions begin with a '5' and are four digits long, if I dialed '5123' on
 the phone, the call would initiate once I pressed the '3'.  Any help would
 be appreciated.

Yes, the phone will do what you want. One of the files the phone downloads
at boot time is dialplan.xml. In it are entries like:
 TEMPLATE MATCH = 0  Timeout=1 User=Phone/
 TEMPLATE MATCH = 3...  Timeout=0 User=Phone/
etc.

I do not have any documents handy that describe the different options, but
the options have much of the same functionality as * exten - statements
have.

Rich


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] SIP debug logs

2004-02-03 Thread James H. Cloos Jr.

 Steve == Steve Foy [EMAIL PROTECTED] writes:

Steve Is there a way of logging all SIP debuging info to a file
Steve somewhere?

Use tethereal or tcpdump to log sip (and/or rtp/rtcp) packets to a
pcap file, then use ethereal (presumably on a different box) to view
them.

-JimC

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

Greg Boehnlein wrote:

 Is this the smallest Asterisk server ever? :)

WHY??? just kidding.  That's pretty cool.  Maybe if you kicked it up to 64
MB, you could create a 4-port sip fxo device and stop all of these posts
about not being able to find one...

This could be good news for the embedded front.

Now, here's the real question: can you install it on a toaster?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Voicetronix Audio Problems when making two or more simultanoues calls

2004-02-03 Thread Peter Zion

Hi David,

I've been working with an * setup with a VoiceTronix 4-port FXO card and
I had similar problems with detecting dialtone.  I had to chan_vpb.c to
use a VPB driver call progress dial command, and to change the dialtone
dectection (I'm in North America) -- please see patch against version
0.7.1 below.  I have not submitted this patch upstream because it's a
workaround that works for my application but leaves lots of problems
unfixed, and possibly creates other problems, so use at your own risk!

I have not yet worked with more than one line with this card so I cannot
say whether I have the other problem you mentioned.

In general, my experience using this card with Asterisk has been
negative.  I hope that Digium will be releasing a multi-port FXO card
soon.

Peter Zion


--BEGIN PATCH--
diff -rpuN ../.build_orig/asterisk/channels/chan_vpb.c
asterisk/channels/chan_vpb.c
--- ../.build_orig/asterisk/channels/chan_vpb.c 2004-01-28
12:36:42.0 -0500
+++ asterisk/channels/chan_vpb.c2004-01-28 12:36:35.0 -0500
@@ -280,17 +280,21 @@ static inline int monitor_handle_owned(s
  break;
 
  case VPB_CALLEND:
- if (e-data == VPB_CALL_CONNECTED)
+ if (e-data == VPB_CALL_CONNECTED ||
+  e-data == VPB_CALL_NO_RING_BACK)
   f.subclass = AST_CONTROL_ANSWER;
- else if (e-data == VPB_CALL_NO_DIAL_TONE ||
-  e-data == VPB_CALL_NO_RING_BACK)
+ else if (e-data == VPB_CALL_NO_DIAL_TONE)
   f.subclass =  AST_CONTROL_CONGESTION;
  else if (e-data == VPB_CALL_NO_ANSWER ||
   e-data == VPB_CALL_BUSY)
   f.subclass = AST_CONTROL_BUSY;
  else if (e-data  == VPB_CALL_DISCONNECTED) 
   f.subclass = AST_CONTROL_HANGUP;
- break;
+
+ if (f.subclass != AST_CONTROL_ANSWER)
+   vpb_sethook_sync(p-handle, VPB_ONHOOK);
+
+ break;
 
  case VPB_STATION_OFFHOOK:
   f.subclass = AST_CONTROL_ANSWER;
@@ -459,8 +463,12 @@ static void *do_monitor(void *unused)
  ast_mutex_lock(monlock),
   ast_mutex_lock(iflock); {
   struct vpb_pvt *p = iflist; /* Find the pvt structure */   
+   int len;

   vpb_translate_event(e, str);
+   len = strlen(str);
+   if (len  0  str[len-1] == '\n')
+str[len-1] = '\0';
   
   if (e.type == VPB_NULL_EVENT) 
goto done; /* Nothing to do, just a wakeup call.*/
@@ -546,7 +554,8 @@ struct vpb_pvt *mkif(int board, int chan
 {
  struct vpb_pvt *tmp;
 
-
+ VPB_DETECT tone;
+ 
  tmp = (struct vpb_pvt *)calloc(1, sizeof *tmp);
 
  if (!tmp)
@@ -560,6 +569,13 @@ struct vpb_pvt *mkif(int board, int chan
  free(tmp);
  return NULL;
  }
+ 
+ vpb_gettonedet(tmp-handle, VPB_DIAL, tone);
+ tone.freq1 = 400;
+ tone.bandwidth1 = 140;
+ tone.glitch = 100;
+ tone.stran[1].tfire = 250;
+ vpb_settonedet(tmp-handle, tone);
   
  if (echocancel) {
  if (option_verbose  4)
@@ -712,11 +728,15 @@ static int vpb_call(struct ast_channel *
 
  vpb_sethook_sync(p-handle,VPB_OFFHOOK);
  
- res = vpb_dial_async(p-handle, s);
+ res = vpb_call_async(p-handle, s);
 
  if (res != VPB_OK) {
ast_log(LOG_DEBUG, Call on %s to %s failed: %s\n, 
ast-name, dest, vpb_strerror(res));  
+   
+   vpb_sethook_sync(p-handle, VPB_ONHOOK);
+   ast_setstate(ast, AST_STATE_DOWN);
+
res = -1;
  } else 
res = 0;
@@ -733,8 +753,6 @@ static int vpb_call(struct ast_channel *
  vpb_timer_start(p-timer);
 }
 p-calling = 1;
-   ast_setstate(ast, AST_STATE_RINGING);
-   ast_queue_control(ast,AST_CONTROL_RINGING, 0);  
 }
 
 return res;
--END PATCH--

On Tue, 2004-02-03 at 00:11, David Liu wrote:
 Hi there,
  
 Besides the problem of Voicetronix dialing too early before the
 carrier gives a dial tone, there also appears to be issues with the
 audio quality when more than 1 channel is utilized.
  
 From an experiment, whenever 1 channel is occupied (i.e. outbound call
 in progress) then getting a dial tone on an available channel will
 take sometime.  What this means is, once you get a dial tone after
 pressing 9, the channel is bridged with the SIP phone, then you wait
 about 4 to 5 seconds and you will hear a dial tone from the carrier. 
 The dial tone is very choppy and most of the time we can't even dial
 out because the quality is so bad that the DTMF is not sent out at
 all.  
  
 It is definitely not the individual line as this scenerio can happen
 at any channel.
  
 Anybody had similiar problems?  
  
 David

___
Asterisk-Users mailing list
[EMAIL PROTECTED]

Re: [Asterisk-Users] busy tones

2004-02-03 Thread Matteo Rancilio

I tried with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+CAPI+readmediff=2 



My extensions are

[outgoing]
exten = 0,1,Goto(outgoing-isdn,s,1)
[outgoing-isdn]
exten = s,1,NoOp()
exten = _X.,1,Dial(CAPI/mynumber:b${EXTEN}|30)
Any ideas?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Jeremy Jones

If asterisk'll compile against uclibc, it'll go on the toaster.  Most
toasters (and coffee grinders  such) don't have enough flash memory for
a full glibc... 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Gomillion
Sent: Tuesday, February 03, 2004 10:01 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

Greg Boehnlein wrote:

 Is this the smallest Asterisk server ever? :)

WHY??? just kidding.  That's pretty cool.  Maybe if you kicked it up to
64
MB, you could create a 4-port sip fxo device and stop all of these posts
about not being able to find one...

This could be good news for the embedded front.

Now, here's the real question: can you install it on a toaster?

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Panny Malialis

Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox.

I was thinking of using that as an IAX-sip translator for offices with NAT.

CPU MPC855T (PowerPC Dual-CPU)
Memory 32MB RAM / 4MB Flash (TS100)
Interfaces1 Ethernet 10/100BT on RJ45
1 RS232 Console on RJ45
RS232 Serial Ports on RJ45

Looks like fun! Although a little lacking on memory.

Any comments?

Panny
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] cisco 7912 voicemail/dnd issue

2004-02-03 Thread Lele Forzani

On Tuesday 03 February 2004 13:10, Swen Veckes wrote:

  In addition it doesn't
  seem you can
  disable (at least on my sip fw release) the
  redirect-to-vm-on-busy feature.

 Actually I think of a soultion like changeing the VM app to be canceld when
 dialing
 * and one can enter the vm and pin to access his messages otherwhise just
 leave
 a message. So I don't need to deal with different VMNo. for one user.
 But no glue how to make it ;(

Better ack and ignore the redirect in chan_sip, and implement vm-on-busy with 
asterisk extension logic. This way you can easily manipulate the original 
extension number and jump to the right mailbox, and keep acd applications, 
that expect busy extensions to be really busy, happy.



bye
lele

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread James Sharp

 Now, here's the real question: can you install it on a toaster?

It builds and runs on NetBSD, minus the hardware part (for the
moment)...so yeah.

Asterisk on NetBSD/Vax.  Hrm.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Playing announcement to called user prior toConfirmation

On Tuesday 03 February 2004 08:57, Matthew B Marlowe wrote:
 I have a Dial Statement and at the end ,m,A(transfer) but when the
 extension picks up it doesn't play anything

Well, that would be why it doesn't work.  Please recheck the help
document.  You will find that you cannot separate options with commas,
as that character is reserved for separating arguments.  Try, instead,
mA(transfer).

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Chris Albertson


Smallest Asterisk server?  No.  That old Gateway box must
be about 2 cubic feet.  1.5 ft^3 at a minimum.  I've got one
that is about 0.2 ft^3 a factor of maybe 10 smaller.

I've installed a working Asterisk server on an older Toshiba
notebok PC.  The Notebook has a 144Mhz Pentium, 80MB RAM
and a 2GB disk.  For very low volume VOIP-only in does OK

Advantages of a Notebook:
  1) Very small, no fan, no noise.
  2) Comes with built-in battry backup
  3) WHat else can you do with a 144Mhz PC?
  4) Can run softphone or Asterisk console phone using
 built-in sound

Disadvangates
  1) How to connect it to the PSTN?

If someone would write a zap driver for the a common PC card
modem (do they still sell these?) then we'd have a realy nice
FXO/VIOP gateway.  Most notebook have two PC card slots

I have an _even older_ Notebook can (and this is the good part)
has a docking staion that has a PCI bus)  So I'm thinking of
putting the digium card(s) in there.  The PC is a 486DX2 at
100Mhz with 16MB RAM.  I couldn't get it to work due to the
16MB RAM but after reading the below maybe I'll try again.


--- Greg Boehnlein [EMAIL PROTECTED] wrote:
 Hello all,
   Saturday night, after a couple of shots of bourbon, I realized 
 that I had an old PC sitting in the garage that I could use as an
 Asterisk 
 gateway if I just blew the dust off it and reloaded it with a modern
 Linux 
 distribution. In my characteristically impulsive manner, I grabbed it
 and 
 started cleaning it up so that I could put it in my office without my
 wife 
 having a fit.
   The sytem is an old Gateway system, that I used to use as an 
 X-terminal. Nothing special really, P-133, 16 megs of ram, 3 PCI
 slots, 
 3.2 gig hard drive. The box booted and I was greated with a RH9 login
 
 screen from my X-server.
   After imaging the hard drive over to my server for backup 
 purposes, I proceeded to try installing Fedora, RH9, RH8 and finally
 RH73 
 without any luck. The 16 megs of ram was just too small to do the 
 installation. So I grabbed a Debian 3.0 netinstall image and got the
 box 
 online and running.
   8 hours later, apt-get dist-upgrade completed and the box was 
 running Debian 3.0 unstable. Now it was time install Asterisk. An 
 apt-cache search asterisk revealed that Debian unstable has pkg
 files 
 available. Yay! That'll save me the time of bulding everything on
 this 
 box so all I will need to do is rebuild the Zaptel modules.
   20 minutes later, I had my Zaptel modules built and was ready to 
 give it a whirl, so I loaded the wcfxo module and started Asterisk.
 My 
 GrandStream registered against the server and I was able to able to
 place 
 calls out the PSTN using the box.
   Initially, I was prepared for this to be an excercise in futility, 
 but I have been extremely surprised by the results. I can support up
 to 3 
 concurrent SIP sessions before I start to get degraded quality, and
 the 
 box appears to be rock solid. I have it registered against our
 production 
 Asterisk server at work over my Cable modem, and my staff can simply
 dial 
 3xxx to ring my extension at home. Voicemail works just fine and with
 the 
 addition of the Asterisk-sounds pkg inbond callers now know that we
 are 
 out Gambling and getting drunk when they call.
 
   Is this the smallest Asterisk server ever? :)
 
 asterisk:~# cat /proc/cpuinfo 
 processor   : 0
 vendor_id   : GenuineIntel
 cpu family  : 5
 model   : 2
 model name  : Pentium 75 - 200
 stepping: 12
 cpu MHz : 132.957
 fdiv_bug: no
 hlt_bug : no
 f00f_bug: yes
 coma_bug: no
 fpu : yes
 fpu_exception   : yes
 cpuid level : 1
 wp  : yes
 flags   : fpu vme de pse tsc msr mce cx8
 bogomips: 265.42
 
 asterisk:~# free
  total   used   free sharedbuffers
 cached
 Mem: 13984  13696288  0   1372   
 868
 -/+ buffers/cache:  11456   2528
 Swap:92728  17316  75412
 
 asterisk:~# ps aux
 USER   PID %CPU %MEM   VSZ  RSS TTY  STAT START   TIME
 COMMAND
 root 1  0.0  0.6  1492   84 ?SFeb02   0:00 init
 [2]   
 root 2  0.0  0.0 00 ?SW   Feb02   0:00
 [keventd]
 root 3  0.0  0.0 00 ?SWN  Feb02   0:00
 [ksoftirqd_CPU0]
 root 4  0.0  0.0 00 ?SW   Feb02   0:14
 [kswapd]
 root 5  0.0  0.0 00 ?SW   Feb02   0:00
 [bdflush]
 root 6  0.0  0.0 00 ?SW   Feb02   0:00
 [kupdated]
 root85  0.0  0.0 00 ?DW   Feb02   0:01
 [kjournald]
 root   292  0.0  1.1  1540  164 ?SFeb02   0:00
 /sbin/syslogd
 root   295  0.0  0.0  21564 ?SFeb02   0:01
 /sbin/klogd
 root   309  0.0  0.0  15200 ?SW   Feb02   0:00
 /usr/sbin/inetd
 root   316  0.0  0.4  3064   56 ?S

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

Steven Critchfield wrote:
 On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
 Steven Critchfield wrote:
 On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
 Hi,

 I'm trying to get my X100P to Dial the following sequence to gain
 access to speed dial numbers on my Norstar PBX that the X100 is
 connected to...
 [snip]
 The problem with your example is that a flash must be executed after
 you have a channel since otherwise there is no offhook event to then
 be toggled by the on then off hook, you just would be off hook.
 Second, look at the documentation for flash and you will see that
 the flash command doesn't accept a argument.

 As to how to actually accomplish what you want, I don't know how.

 since you're connecting an X100P to a Norstar PBX, maybe immediate
 mode would work?  I'm not sure how well that would work with a SIP
 client like you were talking about...

 Not to flame, you but you really need to work on those critical
 reading skills. The example message that was quoted till you trimmed
 it had IAX2 as the VoIP not SIP.


Does it matter?  It's VoIP.  The concept is the same: I didn't know if it
could be done in VoIP, and I was reading a few messages about SIP.  Forgive
me, this will be my last message to the user's list.

 Next, Immediate mode only is of use when you have dialed, or been
 dialed. We are still in the state of not having dialed and therefore
 selected a outbound line.

what about something like this?  NOTE: THIS IS NOT WORKING CODE.  It is an
idea, a concept.  If you want to try it to make it work, then you will have
to build on this.

exten = _*XX,1,Dial(Zap/1/1)  (dials a 1 on the outgoing zap interface,
probably needs a short timeout)
exten = _*XX,2,Flash()
exten = _*XX,3,Dial(Zap/1/${EXTEN})

The flash is probably on the wrong side, as I look at it more closely.  This
will probably send the flash to your VoIP client.  But maybe you could look
into scripting with AGI.


 How often do your speed dials change?  If it's not very often, maybe
 you should recreate it in *.  You could have an extension which
 sends you into a context that asks for the speed dial code.  You
 could then key it in, and it would send you where you want to go.
 Not elegant, but it might be good enough.

 This is probably the only useful suggestion, and most likely the only
 one to work.

And with that, I bid the fair Asterisk-User's list a farewell, at least for
posting.  I will now become one of the countless other leaches who give
nothing back to the community.  It was good to get help, and I tried to help
others out, but I have a lot better things to do than spend my time helping
others only to get flamed every time I turn around.

You need to remember that we're all volunteers.  I will only take it in the
teeth so many times before I say goodbye.

Go ahead and rip me a new one.  Have fun.  Rant, rave, call me stupid.  Tell
me I have no value, and that I contribute nothing.  The more you say it, the
more accurate it becomes.

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] sementation fault with mpg123

2004-02-03 Thread john

I'm still getting a sementation fault with mpg123. I have tried different
parameters creating mp3s the last from cd audio ...
lame -m s --resample 8000  -q 0 -a --cbr -b 32
and several versions of mpg123. I have always created 8000 hz outputs. I've
got other * boxes that don't use moh that have been up for months. This one
crashes every couple of days - the verbose output leading to a crash is
below. Is it just my imagination or has mpg123 always been a pain in the
ass...

What are other mp3 parameters are users using to create mp3s?

John



   -- Stopped music on hold on Parked/[EMAIL PROTECTED]/4ZOMBIE
 == Spawn extension (hc_fxs, 501, 1) exited non-zero on 'Zap/13-1'
   -- Hungup 'Zap/13-1'
   -- Started music on hold, class 'default', on [EMAIL PROTECTED]/4
 == Parked [EMAIL PROTECTED]/4 on 501
Ouch ... error while writing audio data: : Broken pipe
 == Parsing '/etc/asterisk/asterisk.conf': Found
sterisk CVS-01/20/04-17:15:14, Copyright (C) 1999-2001 Linux Support
Services, Inc.
ritten by Mark Spencer [EMAIL PROTECTED]

 == Parsing '/etc/asterisk/logger.conf': Found
sterisk Event Logger Started /var/log/asterisk/event_log
 == Manager registered action Ping
 == Manager registered action Logoff


another snip...

This GDB was configured as i386-redhat-linux-gnu.
Core was generated by `asterisk -vvvfg'.
Program terminated with signal 11, Segmentation fault.
#0  0x0805781d in ?? ()

This e-mail was scanned and found clean by Monroe-Woodbury's Antivirus. 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Smallest server continued...

2004-02-03 Thread toms

This thread got me thinking of other servers that would run asterisk. The
obvious question comes up if Xebian (the xbox version of Debian) would run
as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.

Tom Schaefer
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Asterisk compatibility list

2004-02-03 Thread Paulo Mannheimer

Hi All,

We are compiling an Asterisk interoperability list. 

If you have connected Asterisk to either a PBX or another voice/Voip
device (gateway, gatekeeper, etc ...) please drop me an email. I will
compile it and make it available to the list and on the wiki.

Please make sure to send equipment manufacturer, signaling, protocol,
and whatever else you think is relevant.

Best,

PauloHM


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Dialling Hook Flash on Zaptel

On Tue, 2004-02-03 at 11:33, David Gomillion wrote:
 Steven Critchfield wrote:
  On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
  Steven Critchfield wrote:
  On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
  Hi,
 
  I'm trying to get my X100P to Dial the following sequence to gain
  access to speed dial numbers on my Norstar PBX that the X100 is
  connected to...
  [snip]
  The problem with your example is that a flash must be executed after
  you have a channel since otherwise there is no offhook event to then
  be toggled by the on then off hook, you just would be off hook.
  Second, look at the documentation for flash and you will see that
  the flash command doesn't accept a argument.
 
  As to how to actually accomplish what you want, I don't know how.
 
  since you're connecting an X100P to a Norstar PBX, maybe immediate
  mode would work?  I'm not sure how well that would work with a SIP
  client like you were talking about...
 
  Not to flame, you but you really need to work on those critical
  reading skills. The example message that was quoted till you trimmed
  it had IAX2 as the VoIP not SIP.
 
 
 Does it matter?  It's VoIP.  The concept is the same: I didn't know if it
 could be done in VoIP, and I was reading a few messages about SIP.  Forgive
 me, this will be my last message to the user's list.

It probably doesn't matter other than keeping the answer from being
muddied by technologies not involved. Flash most likely will not work in
VoIP as the kind of information sent to a switch via a flash and then
some other interaction is usually sent OOB in VoIP. 

  Next, Immediate mode only is of use when you have dialed, or been
  dialed. We are still in the state of not having dialed and therefore
  selected a outbound line.
 
 what about something like this?  NOTE: THIS IS NOT WORKING CODE.  It is an
 idea, a concept.  If you want to try it to make it work, then you will have
 to build on this.
 
 exten = _*XX,1,Dial(Zap/1/1)  (dials a 1 on the outgoing zap interface,
 probably needs a short timeout)
 exten = _*XX,2,Flash()
 exten = _*XX,3,Dial(Zap/1/${EXTEN})
 
 The flash is probably on the wrong side, as I look at it more closely.  This
 will probably send the flash to your VoIP client.  But maybe you could look
 into scripting with AGI.

When dial times out in priority 1, you loose the zap channel.
Interesting enough, you may have hit on the correct idea though. The
dialing of the line maybe without any digits will pick up the line. Time
it out, and then do a dial out again this time with real digits. This
would simulate picking up the line, flashing(implied through the hangup
and then new dial) and then dialing.

 
  How often do your speed dials change?  If it's not very often, maybe
  you should recreate it in *.  You could have an extension which
  sends you into a context that asks for the speed dial code.  You
  could then key it in, and it would send you where you want to go.
  Not elegant, but it might be good enough.
 
  This is probably the only useful suggestion, and most likely the only
  one to work.
 
 And with that, I bid the fair Asterisk-User's list a farewell, at least for
 posting.  I will now become one of the countless other leaches who give
 nothing back to the community.  It was good to get help, and I tried to help
 others out, but I have a lot better things to do than spend my time helping
 others only to get flamed every time I turn around.
 
 You need to remember that we're all volunteers.  I will only take it in the
 teeth so many times before I say goodbye.
 
 Go ahead and rip me a new one.  Have fun.  Rant, rave, call me stupid.  Tell
 me I have no value, and that I contribute nothing.  The more you say it, the
 more accurate it becomes.

Dude, you need to take your ego out of the equation here. Like any other
human interaction, if you allow it to hit you in the teeth, you only
loose teeth. As you should notice here, when you distill the problem and
people continue to interact, a solution can usually be found. The
solution above is mostly you, and I have just helped simplify it to a
point it should work. Please continue to post. Eventually we all learn,
and as you learn, you will probably flip to the other side of the
interaction you are complaining about.

-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Chris Albertson


I read a report of Asterisk running on a Microsoft X-Box.
That's kind of a stunt as you could buy a decent PC for
the price of a Linux-capable XBox.  Id's still like to
see Asterisk run on very low-end hardware

The Snom IP phone runs Linux inside?  I assume as Linux
is GPL'd Snom will supply the source code?  It would be
fun to install an Asterisk server in a phone.



--- Panny Malialis [EMAIL PROTECTED] wrote:
 Does anyone have it running on a Cyclades T100 ? same as used for
 ntop/nbox.
 
 I was thinking of using that as an IAX-sip translator for offices
 with NAT.
 
 CPU MPC855T (PowerPC Dual-CPU)
 Memory 32MB RAM / 4MB Flash (TS100)
 Interfaces1 Ethernet 10/100BT on RJ45
 1 RS232 Console on RJ45
 RS232 Serial Ports on RJ45
 
 Looks like fun! Although a little lacking on memory.
 
 Any comments?
 
 Panny
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


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

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Using a Dial Statement with option m and t

When using a dial statement of:

exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m)  

The call is placed with the music on hold and works fine but when I add

exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)

The music on hold will not work

If I use a statement of 

exten =
_NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m,A(soundfile))

The music on hold works but the soundfile doesn't get processed

If I use a statement of

exten =
_NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,A(soundfile),m)

The soundfile will get processed but the music on hold does not play and
the caller hears ringing.

I'm sorry for transmitting in HTML.

I hope this is a better explanation.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Todd
Sent: Tuesday, February 03, 2004 10:48 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Using a Dial Statement with option m and t

When I use option t and m together in the same dial statement the 
music on hold doesn't appear to work.



Is this a normal operation?


1) Please don't post with HTML.  Read the archives for several 
lengthy flamewars over this topic.  Comments as to how I suck because 
I don't like HTML will be ignored (this comment not directed at you, 
Matthew.)

2) Please learn to ask complete questions:

   -Include all relevant lines of your extensions.conf file.
   -Include a small clip of the console output that surrounds the 
event where you have difficulty

I suspect I know what lies at the root of your problem, but I don't 
want to spend my time guessing.  We're happy to help here, but you 
need to give more data so we can point out where you are doing 
something wrong, or if you're not doing something wrong, we need to 
see everything so we can determine if this is a bug.

JT
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] SIP debug logs

2004-02-03 Thread Ejay Hire

When troubleshooting, I'll often 
tcpdump -s 0 -w filename.cap -p host (ipaddressofphone)

To capture the entire contents of all packets from or to
ipaddressofphone non-promiscuously to filename.cap.  Since
my workstation is Win*, I have to sz to move the capture
over to my desktop and then open it in Ethereal.  

This is really helpful when you are working with NAT issues.
I.e. some netgear boxes badger up UDP checksums, and this is
the easy way to see it.

-ejay

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf
Of 
 James H. Cloos Jr.
 Sent: Tuesday, February 03, 2004 10:47 AM
 To: Asterisk-Users
 Subject: Re: [Asterisk-Users] SIP debug logs
 
  Steve == Steve Foy [EMAIL PROTECTED] writes:
 
 Steve Is there a way of logging all SIP debuging info to
a file
 Steve somewhere?
 
 Use tethereal or tcpdump to log sip (and/or rtp/rtcp)
packets to a
 pcap file, then use ethereal (presumably on a different
box) to view
 them.
 
 -JimC
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

On Tue, 2004-02-03 at 12:01, Chris Albertson wrote:
 I read a report of Asterisk running on a Microsoft X-Box.
 That's kind of a stunt as you could buy a decent PC for
 the price of a Linux-capable XBox.  Id's still like to
 see Asterisk run on very low-end hardware
 
 The Snom IP phone runs Linux inside?  I assume as Linux
 is GPL'd Snom will supply the source code?  It would be
 fun to install an Asterisk server in a phone.

Similarly, I know there was a stink about Linksys using linux inside a
router. I just picked up a USR 802.11g router that would be cool to get
a small VoIP only asterisk install on. It would make setting up those
802.11b phones nice and easy.
-- 
Steven Critchfield  [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] x100p card conflicts with DSL modem

2004-02-03 Thread Michael Zheng

Hi, all

When I use x100p card, my DSL modem can not connect
with ISP. Is my card bad or all x100p conflict with
DSL modem?

Best,
Michael

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Smallest server continued...

2004-02-03 Thread Chris Albertson


--- [EMAIL PROTECTED] wrote:
 This thread got me thinking of other servers that would run asterisk.
 The
 obvious question comes up if Xebian (the xbox version of Debian)
 would run
 as a SIP only server? Asterisk on an XBox would be a small box! Cheap
 too.

It's been done.  In fact by Mark hiom self if you beleive this
URL that Goole found. 
http://216.239.53.104/search?q=cache:M1pPrvOlBewJ:nlug.org/mail/nlug__2003_12/0094.html+linux+asterisk+xboxhl=enie=UTF-8

But in my opinion Asterisk running on a Snom 100 would be even cooler
and I can think of uses for it already.


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

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread listas iPfone

Snom Does gives the souce and more:

http://www.snom.com/sources_en.php

- Original Message - 
From: Chris Albertson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 4:01 PM
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

 I read a report of Asterisk running on a Microsoft X-Box.
 That's kind of a stunt as you could buy a decent PC for
 the price of a Linux-capable XBox.  Id's still like to
 see Asterisk run on very low-end hardware

 The Snom IP phone runs Linux inside?  I assume as Linux
 is GPL'd Snom will supply the source code?  It would be
 fun to install an Asterisk server in a phone.

 --- Panny Malialis [EMAIL PROTECTED] wrote:
  Does anyone have it running on a Cyclades T100 ? same as used for
  ntop/nbox.

  I was thinking of using that as an IAX-sip translator for offices
  with NAT.

  CPU MPC855T (PowerPC Dual-CPU)
  Memory 32MB RAM / 4MB Flash (TS100)
  Interfaces1 Ethernet 10/100BT on RJ45
  1 RS232 Console on RJ45
  RS232 Serial Ports on RJ45

  Looks like fun! Although a little lacking on memory.

  Any comments?

  Panny
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users

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

 __
 Do you Yahoo!?
 Yahoo! SiteBuilder - Free web site building tool. Try it!
 http://webhosting.yahoo.com/ps/sb/
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Using a Dial Statement with option m and t

2004-02-03 Thread William Waites

On Tue, Feb 03, 2004 at 01:04:27PM -0500, Matthew B Marlowe wrote:
 
 exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)
 
 The music on hold will not work

I believe you do not want a comma between the t and the m.

-w
-- 
/~\  The ASCII Ribbon Campaign
\ /No HTML/RTF in email
 X No Word docs in email
/ \  Respect for open standards
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Using a Dial Statement with option m and t

2004-02-03 Thread Eric Wieling

You do not put a , between t,m or any of the end parameters.

See show application dial

On Tue, 2004-02-03 at 12:04, Matthew B Marlowe wrote:
 When using a dial statement of:
 
 exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m)  
 
 The call is placed with the music on hold and works fine but when I add
 
 exten = _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,t,m)
 
 The music on hold will not work
 
 If I use a statement of 
 
 exten =
 _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,m,A(soundfile))
 
 The music on hold works but the soundfile doesn't get processed
 
 If I use a statement of
 
 exten =
 _NXXNXX,6,Dial(SIP/611SIP/612SIP/613SIP/614,30,A(soundfile),m)
 
 The soundfile will get processed but the music on hold does not play and
 the caller hears ringing.
 
 I'm sorry for transmitting in HTML.
 
 I hope this is a better explanation.
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of John Todd
 Sent: Tuesday, February 03, 2004 10:48 AM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Using a Dial Statement with option m and t
 
 When I use option t and m together in the same dial statement the 
 music on hold doesn't appear to work.
 
 
 
 Is this a normal operation?
 
 
 1) Please don't post with HTML.  Read the archives for several 
 lengthy flamewars over this topic.  Comments as to how I suck because 
 I don't like HTML will be ignored (this comment not directed at you, 
 Matthew.)
 
 2) Please learn to ask complete questions:
 
-Include all relevant lines of your extensions.conf file.
-Include a small clip of the console output that surrounds the 
 event where you have difficulty
 
 I suspect I know what lies at the root of your problem, but I don't 
 want to spend my time guessing.  We're happy to help here, but you 
 need to give more data so we can point out where you are doing 
 something wrong, or if you're not doing something wrong, we need to 
 see everything so we can determine if this is a bug.
 
 JT
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
Asterisk Resource Pages.

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

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Smallest server continued...

2004-02-03 Thread William Waites

On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote:
 This thread got me thinking of other servers that would run asterisk. The
 obvious question comes up if Xebian (the xbox version of Debian) would run
 as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.

I see no reason you couldn't run it on some of the
handheld pcs... Perhaps one with audio hardware and
wireless ethernet... It'd make a great softphone...

/w
-- 
/~\  The ASCII Ribbon Campaign
\ /No HTML/RTF in email
 X No Word docs in email
/ \  Respect for open standards
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Tony Kava

  The Snom IP phone runs Linux inside?  I assume as Linux
  is GPL'd Snom will supply the source code?  It would be
  fun to install an Asterisk server in a phone.
 
 Similarly, I know there was a stink about Linksys using linux 
 inside a router. I just picked up a USR 802.11g router that 
 would be cool to get a small VoIP only asterisk install on. 
 It would make setting up those 802.11b phones nice and easy.

That is a neat idea.  I think the Linksys product you mentioned (or at least
one Linux-based Linksys product) is the WRT54G.  I believe Linksys took
steps with their latest firmware to prevent people from messing around
inside the router.  Below this paragraph is a link to someone's
informational site on the model.  They were running snort IDS on the unit.
If someone has one of these units perhaps they could let us know if they get
anywhere with an Asterisk install. :)

http://www.batbox.org/wrt54g.html

--
Tony Kava
Senior Network Administrator
Pottawattamie County, Iowa


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Christian Stredicke

Well I also though about this five minutes ago... I think the biggest
problem should be memory (we have 16 MB DRAM and 4 MB Flash). 

Also, the question is if the plastic makes a box impression...

Christian

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of listas iPfone
 Sent: Tuesday, February 03, 2004 7:16 PM
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
 
 Snom Does gives the souce and more:
 
 http://www.snom.com/sources_en.php
 
 - Original Message -
 From: Chris Albertson [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, February 03, 2004 4:01 PM
 Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
 
 
 
  I read a report of Asterisk running on a Microsoft X-Box.
  That's kind of a stunt as you could buy a decent PC for
  the price of a Linux-capable XBox.  Id's still like to
  see Asterisk run on very low-end hardware
 
  The Snom IP phone runs Linux inside?  I assume as Linux
  is GPL'd Snom will supply the source code?  It would be
  fun to install an Asterisk server in a phone.
 
 
 
  --- Panny Malialis [EMAIL PROTECTED] wrote:
   Does anyone have it running on a Cyclades T100 ? same as used for
   ntop/nbox.
  
   I was thinking of using that as an IAX-sip translator for offices
   with NAT.
  
   CPU MPC855T (PowerPC Dual-CPU)
   Memory 32MB RAM / 4MB Flash (TS100)
   Interfaces1 Ethernet 10/100BT on RJ45
   1 RS232 Console on RJ45
   RS232 Serial Ports on RJ45
  
   Looks like fun! Although a little lacking on memory.
  
   Any comments?
  
   Panny
   ___
   Asterisk-Users mailing list
   [EMAIL PROTECTED]
   http://lists.digium.com/mailman/listinfo/asterisk-users
   To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
  =
  Chris Albertson
Home:   310-376-1029  [EMAIL PROTECTED]
Cell:   310-990-7550
Office: 310-336-5189  [EMAIL PROTECTED]
KG6OMK
 
  __
  Do you Yahoo!?
  Yahoo! SiteBuilder - Free web site building tool. Try it!
  http://webhosting.yahoo.com/ps/sb/
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: sementation fault with mpg123

2004-02-03 Thread James H. Cloos Jr.

| I'm still getting a sementation fault with mpg123. 

Isn't it time to get mg3 out of the equation?

Sox can convert just about anything to 16 bit signed mono pcm in
just about any container that support that.  It looks like *'s
format_wav.c is for exactly that format, so for local files we
should be running:

   sox $FOO.$BAR -s -w -c 1 -r 8000 $FOO.wav resample -ql

for any $FOO and any $BAR and using those files for moh.

Then * can handle it all itself.

(Alternatively, use -g instead of -s -w and save a .gsm file,
just like everything else in /var/lib/asterisk/sounds.)

-JimC

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] x100p card conflicts with DSL modem

2004-02-03 Thread Dave Cotton

On Tue, 2004-02-03 at 19:14, Michael Zheng wrote:
 Hi, all
 
 When I use x100p card, my DSL modem can not connect
 with ISP. Is my card bad or all x100p conflict with
 DSL modem?

I have an X100p working in the same box as a Bewan PCI ADSL modem with
no problems. But adding a radio card caused no end of problems.

-- 
Dave Cotton [EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Smallest server continued...

2004-02-03 Thread Florian Overkamp

Hi,

 -Original Message-
 This thread got me thinking of other servers that would run 
 asterisk. The obvious question comes up if Xebian (the xbox 
 version of Debian) would run as a SIP only server? Asterisk 
 on an XBox would be a small box! Cheap too.

The Xbox has USB ports, right ? You could connect some S100U's maybe. I have
something in the back of my mind saying Mark has done this in the past.

Grt,
Florian


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Re: sementation fault with mpg123

2004-02-03 Thread mattf

I'd love to have a non-mp3 music-on-hold option. Anybody put this as a
feature request yet?

MATT---

-Original Message-
From: James H. Cloos Jr. [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 1:42 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: sementation fault with mpg123

| I'm still getting a sementation fault with mpg123. 

Isn't it time to get mg3 out of the equation?

Sox can convert just about anything to 16 bit signed mono pcm in
just about any container that support that.  It looks like *'s
format_wav.c is for exactly that format, so for local files we
should be running:

   sox $FOO.$BAR -s -w -c 1 -r 8000 $FOO.wav resample -ql

for any $FOO and any $BAR and using those files for moh.

Then * can handle it all itself.

(Alternatively, use -g instead of -s -w and save a .gsm file,
just like everything else in /var/lib/asterisk/sounds.)

-JimC

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] x100p card conflicts with DSL modem

On Tuesday 03 February 2004 12:14, Michael Zheng wrote:
 When I use x100p card, my DSL modem can not connect
 with ISP. Is my card bad or all x100p conflict with
 DSL modem?

Perhaps you forgot to put a filter between the line and your X100P?

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] sementation fault with mpg123

On Tuesday 03 February 2004 11:46, john wrote:
 I'm still getting a sementation fault with mpg123. I have tried

Ah, adventures in the pubic school system.

 This GDB was configured as i386-redhat-linux-gnu.
 Core was generated by `asterisk -vvvfg'.
 Program terminated with signal 11, Segmentation fault.
 #0  0x0805781d in ?? ()

Can you give us a 'bt full' at this point?

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] x100p card conflicts with DSL modem

2004-02-03 Thread Eric Wieling

Do you have a DSL filter on your X100P?  Just like any other telephone
device it needs a DSL filter to keep it from messing up your DSL
service.  

On Tue, 2004-02-03 at 12:14, Michael Zheng wrote:
 Hi, all
 
 When I use x100p card, my DSL modem can not connect
 with ISP. Is my card bad or all x100p conflict with
 DSL modem?
 
 Best,
 Michael
 
 __
 Do you Yahoo!?
 Yahoo! SiteBuilder - Free web site building tool. Try it!
 http://webhosting.yahoo.com/ps/sb/
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Go to http://www.digium.com/index.php?menu=documentation and look at
the Unofficial Links section.  This section has links to a wide
variety of 3rd party Asterisk related pages.  My page is the
Asterisk Resource Pages.

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

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Smallest server continued...

2004-02-03 Thread Brian Johnson

linphone is available for ipaqs running familiar/opie linux.

It's on my todo list to try it out via wifi



William Waites ([EMAIL PROTECTED]) wrote:

 On Tue, Feb 03, 2004 at 10:46:50AM -0700, [EMAIL PROTECTED] wrote:
  This thread got me thinking of other servers that would run asterisk. The
  obvious question comes up if Xebian (the xbox version of Debian) would run
  as a SIP only server? Asterisk on an XBox would be a small box! Cheap too.

 I see no reason you couldn't run it on some of the
 handheld pcs... Perhaps one with audio hardware and
 wireless ethernet... It'd make a great softphone...

 /w
 --
 /~\  The ASCII Ribbon Campaign
 \ /No HTML/RTF in email
  X No Word docs in email
 / \  Respect for open standards
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Still looking for small fxo sip gateway

2004-02-03 Thread Clif Jones

Comments below.

Rich Adamson wrote:

I've been looking around for a small external sip fxo gateway, sending
emails to possible vendors, etc, and can not seem to come up with anything
that fits. Suggestions anyone? (No channel bank  T1 card suggestions, 
please. I've also just completed an eval of the Mediatrix 1204 which
does not support the requirements.)
 

I second that.  Mediatrix is not RFC3261 compliant.

The market between two fxo pstn lines (pair of x100p's) and something
around four to six lines seems to be lacking, or I'm looking in the
wrong search engine (or something). I fully understand the economics of
when a channel bank and T1 card becomes cost effective, including the 
eBay costs (and risks), etc. I've also heard the comments for months 
now that Digium is/will be selling something real-soon-now.
 

I'm in the process of doing the same thing for a friend's business that 
I have installed * for their
PBX solution.  I work for a company that develops SIP based technologies 
for carriers so the
gateways are extremely expensive but work.  What I have found is that 
these SOHO type
gateways are very unstable, non-standard and/or lacking in basic features.

Specifically, I'd like to use a 4-port fxo sip gateway capable of supporting
four US pstn analog lines, CallerID, Touchtone, loop style supervision,
and have the capability for asterisk to direct an outbound call to a 
specific port on that gateway. I think that implies each port must
execute a sip register command successfully. It's also expected to accept 
incoming pstn calls directing those to a single asterisk. (I don't care 
about an IP dialtone, nat, etc, just a plain-jane two-way sip gateway.)
 

The closest thing to your requirements that I am working with is an 
Audiocodes MP-104 or larger
gateway.  My friend bought a MP-104 and it has 4-ports that we have 
configured for FXO.  It
has Caller-ID, hunt-groups in any combination of the ports.  The only 
problem that I am having
with it is DTMF relay, which I will hopefully resolve with their latest 
firmware load.

SIP gateways normally do NOT register.  Some smaller ones may but this 
does not scale.  Imagine
a bunch of 24-port FXO gateways all registering at once!  You normally 
just set the proxy on the
gateway, give each port an ID and on the PBX/proxy have a routing rule 
that goes out via the
gateway.

If anyone is designing such a box and need professional eval, we can 
certainly work with you privately (off list to radamson @ routers dot com)
to accomidate those needs.

Anyone seen such a beast at a reasonable price?
 

I think the Audiocodes MP-104 cost was around $1200 last year.

Rich



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Qualify statement

2004-02-03 Thread Senad Jordanovic

Does anyone know, is there a way to get current status of device
From * using some variable or similar in relation to qualify=XXX
statement.

I am referring to qualify= which qualifies and monitors if device
is reachable.

I need this in order to include it in my dial plan so that incoming call
can be redirected elsewhere if device is not accessible.

Ta
SJ


___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread James Sharp

 On Tue, 3 Feb 2004, Chris Albertson wrote:

 Smallest Asterisk server?  No.  That old Gateway box must
 be about 2 cubic feet.  1.5 ft^3 at a minimum.  I've got one
 that is about 0.2 ft^3 a factor of maybe 10 smaller.

 Hehehe.. As far as Form Factor goes, I'm sure there are smaller boxes
 out there. How about Most resource challenged Asterisk server ever? :)


How about one of the 1Ghz Soekris boards with a 802.11 board in it?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Greg Boehnlein

On Tue, 3 Feb 2004, Steven Critchfield wrote:

 On Tue, 2004-02-03 at 12:01, Chris Albertson wrote:
  I read a report of Asterisk running on a Microsoft X-Box.
  That's kind of a stunt as you could buy a decent PC for
  the price of a Linux-capable XBox.  Id's still like to
  see Asterisk run on very low-end hardware
  
  The Snom IP phone runs Linux inside?  I assume as Linux
  is GPL'd Snom will supply the source code?  It would be
  fun to install an Asterisk server in a phone.
 
 Similarly, I know there was a stink about Linksys using linux inside a
 router. I just picked up a USR 802.11g router that would be cool to get
 a small VoIP only asterisk install on. It would make setting up those
 802.11b phones nice and easy.

I think this would be a stretch. I've done quite a bit of hacking on 
uClinux and embedded systems (http://myturl.com/0009L) and the lack of a 
MMU and some of the standard stuff that every PC based Linux user is used 
to can be a pretty difficult road to trod.

Don't get me wrong, I'm all for trying, but running it on the WR54g and/or 
the Actiontec Dual PC modem is probably not an option.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RedHat 9 VSFTPD Digium Hardware Oddoties

2004-02-03 Thread Mindworks Wireless

Here is my experience so far to treat some issues I have been having with
Digium hardware (t100p, and x100p's.)  I am not 100% certain these are
fixxes, but just something for people to try if they are expierencing
issues with the hardware performing quirky.

1st)  Do NOT use Promise Array ATA Raid controllers in a sytem with Digium
Hardware.  This created many random red alarm issues with the X100P card,
as I would see an error generated by both Asterisk and the Array Card.

2.)  RedHat 9 with vsftpd running, set tcpwrapper=yes if you use the
Asterisk Box as a FTP boot server and the PBX.  My polycom phones would
sometime act quirky, and I would get multiple errors from vsftpd about a
bad file descriptor.  The exact error messages would be:

Feb  1 10:17:34 dunpbx vsftpd: warning: can't get client address:
Bad file descriptor
Feb  1 13:51:49 dunpbx last message repeated 2 times
Feb  1 16:37:06 dunpbx last message repeated 4 times
Feb  1 17:06:01 dunpbx last message repeated 2 times
Feb  1 17:07:15 dunpbx last message repeated 2 times
Feb  1 20:08:11 dunpbx last message repeated 2 times
Feb  2 07:51:11 dunpbx last message repeated 2 times
Feb  2 10:29:03 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 10:31:18 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 12:17:14 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 12:33:22 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 15:17:27 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 15:22:47 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor
Feb  2 15:47:48 dunpbx vsftpd: warning: can't get client address: Bad file
descriptor

Setting to tcpwrapper=yes in the vsftpd.conf resolves this issue.  I
noticed that when I would see a slew of these messages, I would see random
red alarms on my PBX:

Feb  3 03:16:07 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 03:16:09 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 03:16:18 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 03:16:30 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:14:24 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:14:26 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:14:36 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:14:42 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:14:47 NOTICE[1167272000]: Fax detected, but no fax extension
Feb  3 04:14:48 WARNING[1167272000]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:14:51 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:14:57 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:14:59 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:15:08 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:15:11 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 04:15:20 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 04:15:30 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:18:45 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:18:47 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:18:57 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:19:04 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:19:09 NOTICE[1167272000]: Fax detected, but no fax extension
Feb  3 05:19:10 WARNING[1167272000]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:19:13 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:19:18 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:19:22 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:19:29 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:19:33 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:19:40 WARNING[1133718080]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:20:04 NOTICE[1133718080]: Alarm cleared on channel 2
Feb  3 05:20:09 WARNING[1167272000]: Detected alarm on channel 2: Red
Alarm
Feb  3 05:20:13 NOTICE[1133718080]: Alarm cleared on channel 2

Just some things I wanted to throw out there in case anyone else runs into
something weird and this might help them.  Again, I am not one hundred
percent certain this resolves the issues above, but it certainly seems
like it has helped.

Regards,

Brent

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Mediatrix sip fxo gateway workaround?

Possible Mediatrix 1204 fxo sip gateway workaround

Need some feedback from experienced * users relative to this workaround
please please please.

Problem: The mediatrix 4-port fxo gateway does not provide any mechanism
for * to select which port an outbound pstn call will use. (See lots
of previous posts over the past four days for more detail if needed.)

Our reseller has been working with Mediatrix to find a way for *
to send pstn calls to a specific port number on their 1204 4-port fxo sip 
gateway. The proposed work around (below) sets a unique-but-well-known 
CallerID prior to sending the call to the 1204, and the 1204 filters on 
the CallerID sending the outbound call to the designated port. (The 1204
_does_ have such filtering/routing capability.)

Since this unique callerid is _never_ forwarded to the US pstn providers,
does anyone see any technical or management problem with using this approach
both in the short and long term???

I'm thinking this is an acceptable workaround since it does not require
micro-managing the dialplan, the 1204, etc. In my case, I'm not very concerned
with scaling the solution since we could only hope business would increase
to the point where four additional pstn analog lines were needed. ;)
(FWIW, a 3,congestion statement can be added to the proposed statements.)

Thoughts anyone?
Rich

 Ok, you need to use the net2pstnsourcefilter  to make this work. In this
 example you need to set port 1 to , 2 to , 3 to , 4 to  .
 Then with the extension configuration below, and number starting with 9 will
 go to port 1 with the 9 removed from the string sent. Any number starting
 with 8 will be sent to port 2 with the 8 removed from the string sent and so
 on. It works like a charm on my 1204.
 
 [SIP]
 exten = _9.,1,SETCIDNUM()
 exten = _9.,2,Dial,SIP/[EMAIL PROTECTED]
 
 exten = _8.,1,SETCIDNUM()
 exten = _8.,1,Dial,SIP/[EMAIL PROTECTED]
 
 exten = _7.,1,SETCIDNUM()
 exten = _7.,2,Dial,SIP/[EMAIL PROTECTED]
 
 exten = _6.,1,SETCIDNUM()
 exten = _6.,2,Dial,SIP/[EMAIL PROTECTED]



___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread James H. Thompson

Check here for list of small Asterisk implementations mentioned on the mailing list.

http://www.voip-info.org/wiki-Asterisk+setup+minimum

Jim

James H. Thompson
[EMAIL PROTECTED]

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread David J Carter

Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for
that.

The Linux bit is all free, and only a couple of PCB work to disenable the
protection.

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris
Albertson
Sent: 03 February 2004 18:01
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?



I read a report of Asterisk running on a Microsoft X-Box.
That's kind of a stunt as you could buy a decent PC for
the price of a Linux-capable XBox.  Id's still like to
see Asterisk run on very low-end hardware

The Snom IP phone runs Linux inside?  I assume as Linux
is GPL'd Snom will supply the source code?  It would be
fun to install an Asterisk server in a phone.



--- Panny Malialis [EMAIL PROTECTED] wrote:
 Does anyone have it running on a Cyclades T100 ? same as used for
 ntop/nbox.

 I was thinking of using that as an IAX-sip translator for offices
 with NAT.

 CPU MPC855T (PowerPC Dual-CPU)
 Memory 32MB RAM / 4MB Flash (TS100)
 Interfaces1 Ethernet 10/100BT on RJ45
 1 RS232 Console on RJ45
 RS232 Serial Ports on RJ45

 Looks like fun! Although a little lacking on memory.

 Any comments?

 Panny
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


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

__
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building tool. Try it!
http://webhosting.yahoo.com/ps/sb/
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] New Zealand users/contractors

2004-02-03 Thread matt

Yes...myself.  I can be contacted at the email above or on (021) 1387245.

Kind regards,

Matt Riddell

Are there any New Zealand Asterisk users/contractors out there - we're 
looking to install a small business pnx and are interested in Asterisk 
as a solution.




___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: RE: [Asterisk-Users] cdr mysql problem

2004-02-03 Thread Tomica Crnek


Thanks, I don't know what is different from all steps I have followed
several times. I did all this before, believe me. Now, I said to myself
that I'll do it once again, and it worked. 

Thanks once again!

Tomica

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dipak0105
Sent: Tuesday, February 03, 2004 1:47 PM
To: [EMAIL PROTECTED]
Subject: Re: RE: [Asterisk-Users] cdr mysql problem

Hi 

You have to follow this steps and obviously you got success because we
have followed this steps and got success.

Configure the cdr_mysql.conf file in the location /etc/asterisk. The
configuration is as follows:

[global]
hostname=localhost  ;This is the host name of MySql

dbname=asteriskcdrdb;This is the database name of MySql
password=password   ;This is the password of user
user=asteriskcdruser;This is the user in MySql
port=3306   ;This is the port number running the
MySql
sock=/var/lib/mysql/mysql.sock  ;This is the socket of MySql and the
location

Then follow the steps in the MySql server

1. Go to Start ApplicationSystemService Configuration. Check the
mysqld box. Click the save button
   Then click the start button.

2. Open a kolsole.

3. Type: mysql -u root to enter into mysql.

4. Type: SET PASSWORD FOR root = PASSWORD(password); for setting the
root password.

5. Type: GRANT ALL PRIVILEGES ON *.* TO [EMAIL PROTECTED]
IDENTIFIED BY password WITH GRANT OPTION;

6. Type: \q; to Quit mysql.

7. Type: In konsole mysql -u asteriskcdruser -p and use the password
password,
   to reenter mysql as asteriskcdruser user.

8. Type: CREATE DATABASE asteriskcdrdb;

9. Type:

  CREATE TABLE cdr (
  calldate datetime NOT NULL default '-00-00 00:00:00',
  clid varchar(45) NOT NULL default '',
  src varchar(45) NOT NULL default '',
  dst varchar(45) NOT NULL default '',
  dcontext varchar(45) NOT NULL default '',
  channel varchar(45) NOT NULL default '',
  dstchannel varchar(45) NOT NULL default '',
  lastapp varchar(45) NOT NULL default '',
  lastdata varchar(45) NOT NULL default '',
  duration int(11) NOT NULL default '0',
  billsec int(11) NOT NULL default '0',
  disposition varchar(45) NOT NULL default '',
  amaflags int(11) NOT NULL default '0',
  accountcode varchar(45) NOT NULL default '',
  uniqueid varchar(45) NOT NULL default ''
  );

  for create a table cdr.

10. To reload the configuration, type reload from the Asterisk command
prompt.

These are the steps of Configuration of MySql with Asterisk server.
Contact me if you need any further clarifications.

Dipak Kumar Paul.
Sigmabit Technology India.


[EMAIL PROTECTED] wrote:

Hi, here it is... 

[EMAIL PROTECTED] asterisk]# cat cdr_mysql.conf ; ; Note - if the database
server is hosted on the same machine as the ; asterisk server, you can
achieve a local Unix socket connection by ; setting hostname=localhost ;
; port and sock are both optional parameters.  If hostname is specified
; and is not localhost, then cdr_mysql will attempt to connect to the
; port specified or use the default port.  If hostname is not specified
; or if hostname is localhost, then cdr_mysql will attempt to connect
; to the socket file specified by sock or otherwise use the default
socket ; file.
;
[global]
hostname=localhost
dbname=asteriskcdrdb
password=**
user=asteriskcdruser
;port=3306
;sock=/tmp/mysql.sock
sock=/var/lib/mysql/mysql.sock


srwxrwxrwx1 mysqlmysql   0 Feb  2 19:37
/var/lib/mysql/mysql.sock


Tomica

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Tuesday, February 03, 2004 12:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] cdr mysql problem

On Monday 02 February 2004 15:27, Tomica Crnek wrote:
 Yes, I have checked the logs. There is nothing there. I think asterisk

 doesn't try to connect.

Please paste the contents of /etc/asterisk/cdr_mysql.conf.  Also, paste
the output of:

ls -l /tmp/mysql.sock /var/lib/mysql/mysql.sock ; locate mysql.sock

-Tilghman

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Get Your Private, Free E-mail from Indiatimes at
http://email.indiatimes.com

 Buy The Best In BOOKS at http://www.bestsellers.indiatimes.com

Bid for for Air Tickets @ Re.1 on Air Sahara Flights. Just log on to
http://airsahara.indiatimes.com and Bid Now!

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

Re: [Asterisk-Users] The Smallest Asterisk Server Ever?

2004-02-03 Thread Panny Malialis

I cant wait to see the asterisk on an xbox page!!, but the link seems broken

http://nlug.org/mail/nlugb2003_12/0094.html

I've tried removing the b with no luck

Anyone know what the link should be ?

Thanks

Panny

- Original Message - 
From: David J Carter [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, February 03, 2004 8:31 PM
Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever?


 Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for
 that.
 
 The Linux bit is all free, and only a couple of PCB work to disenable the
 protection.
 
 Dave
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Chris
 Albertson
 Sent: 03 February 2004 18:01
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
 
 
 
 I read a report of Asterisk running on a Microsoft X-Box.
 That's kind of a stunt as you could buy a decent PC for
 the price of a Linux-capable XBox.  Id's still like to
 see Asterisk run on very low-end hardware
 
 The Snom IP phone runs Linux inside?  I assume as Linux
 is GPL'd Snom will supply the source code?  It would be
 fun to install an Asterisk server in a phone.
 
 
 
 --- Panny Malialis [EMAIL PROTECTED] wrote:
  Does anyone have it running on a Cyclades T100 ? same as used for
  ntop/nbox.
 
  I was thinking of using that as an IAX-sip translator for offices
  with NAT.
 
  CPU MPC855T (PowerPC Dual-CPU)
  Memory 32MB RAM / 4MB Flash (TS100)
  Interfaces1 Ethernet 10/100BT on RJ45
  1 RS232 Console on RJ45
  RS232 Serial Ports on RJ45
 
  Looks like fun! Although a little lacking on memory.
 
  Any comments?
 
  Panny
  ___
  Asterisk-Users mailing list
  [EMAIL PROTECTED]
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 =
 Chris Albertson
   Home:   310-376-1029  [EMAIL PROTECTED]
   Cell:   310-990-7550
   Office: 310-336-5189  [EMAIL PROTECTED]
   KG6OMK
 
 __
 Do you Yahoo!?
 Yahoo! SiteBuilder - Free web site building tool. Try it!
 http://webhosting.yahoo.com/ps/sb/
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ___
 Asterisk-Users mailing list
 [EMAIL PROTECTED]
 http://lists.digium.com/mailman/listinfo/asterisk-users
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Still looking for small fxo sip gateway