[asterisk-users] Problems with bridging data calls over Wildcard TE405P

2006-12-06 Thread Marc Rohlfing

 Hi,

I recently installed an Asterisk server at a customer's here in
Germany, using a Wildcard TE405P - Ports 1 and 2 (group 1) are
connected to the phone system (Deutsche Telekom), whereas ports 3 and
4 (group 2) connect to the existing HiPath 3500 PABX.

Currently, the dialplan is doing nothing but routing calls through
(bridging) for the time being:
--
[inbound]
exten = _4048.,1,Dial(Zap/g2/${EXTEN})

[outbound]
exten = _0.,1,SetCIDNum(4048)
exten = _0.,n,Dial(Zap/g1/${EXTEN})
--
This works great for all voice calls inbound and outbound, also faxes
and modem calls are routed just fine.
Data calls from ISDN devices in both directions, however (like home
banking) are only connected, yet no data flows and after a couple
seconds the answering side hangs up.
At the console this looks like
--
   -- Executing SetCIDNum(Zap/119-1, 4048) in new stack
   -- Executing Dial(Zap/119-1, Zap/g1/0531590||tT) in new stack
   -- Requested transfer capability: 0x08 - DIGITAL
   -- Called g1/0531590
   -- Zap/4-1 is ringing
   -- Zap/4-1 answered Zap/119-1
   -- Hungup 'Zap/4-1'
--

I tried to read up on everything regarding data calls, but only found
so much different, sometimes conflicting, info that I'm hoping
somebody here has some experience with that.

 Thanks

   Marc
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[Asterisk-Users] Putting a call recording into a mailbox

2006-07-04 Thread Marc Rohlfing
  Hi,

I was wondering, after recording a call (through either the
monitor()-application or automon), is there a way to put the recorded
file into a user's mailbox? So far, we just send out the file as an
email attachment, but having it in my mailbox would just be so much more
convenient... (^_^)

  Marc Rohlfing

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AW: [Asterisk-Users] Putting a call recording into a mailbox

2006-07-04 Thread Marc Rohlfing
  Hi,

 I don't understand what you are asking: what's the difference 
 between sending out the email as an attachment so it ends up 
 in a user's mailbox versus having it in the user's mailbox. 
 Aren't they the same?

Oops, my bad: I'm talking about the user's *voicemail* box here - should
have been more precise there... 

  Marc Rohlfing

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AW: [Asterisk-Users] syntax error

2006-06-21 Thread Marc Rohlfing
  Hi,

 Does anyone know why this row:
  exten = s,2,GotoIf($[${CALLERIDNAME:0:${LEN(${RGPREFIX})}} !=
 ${RGPREFIX}]?4:3)

took me some squinting, but the parantheses seem correct - so I presume
the Asterisk parser can't cope with that convoluted an expression (using
a function within a variable, basically).
Try putting LEN(${RGPREFIX}) into a separate variable first, then refer
to it in your GotoIf in a second statement.

  Marc

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AW: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06

2006-06-14 Thread Marc Rohlfing
  Good morning,

 Why still use mpg123?
 Start using format_mp3 from asterisk-addons and your * will 
 play mp3 by itself...

good point - did that, and everything's working again. Thanks!

Marc Rohlfing

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[Asterisk-Users] Eicon Diva Server with v3.0 drivers

2006-06-14 Thread Marc Rohlfing
  Hi,

I'm trying to get an Eicon Diva Server4BRI card running under Ubuntu
6.06 - by downloading the v3 driver package from Melware and compiling
everything. Yet, after activating the necessary modules (divas and
divadidd) and interactively configuring the card
(/usr/lib/divas/Config), starting up the adapter fails!

The error /usr/lib/divas/divactrl load -c 1 -Debug produces complains
about the missing protocol image
  A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0)

I know from version 2.0 that you had to download these images from
either isdn4linux.org or melware.de, yet none of them still have the
files available. /usr/lib/divas/ does contain some *etsi* files - do
they help somehow?

Any hint appreciated.

  Marc Rohlfing

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AW: [Asterisk-Users] Eicon Diva Server with v3.0 drivers

2006-06-14 Thread Marc Rohlfing
  Hi,

  The error /usr/lib/divas/divactrl load -c 1 -Debug produces 
  complains about the missing protocol image
A: can't open protocol image: (/usr/lib/divas/te_etsi.2q0)
 It is not necessary any more to download any firmware files 
 (they are incompatible anyway). All needed files are part of 
 the v3 package.
 It is odd, that the file te_etsi.* is searched. This is 
 needed only if the DMLT code (te_dmlt.*) is not available.
 Can you please provide the list of files which are installed 
 in /usr/lib/divas and possible logs /var/log/diva* ?

As usual, the second I sent my request, I tried something else and it
worked (^_^)

Seriously: If I run the autogenerated startup script
(/usr/lib/divas/divas_cfg.rc), the card is activated just fine. capiinfo
shows all 8 B-channels, so I guess I'm good to go.
Maybe this should be stated more clearly in the INSTALL and README files
- it's especially confusing for veterans who try to do things the 2.1
way...

In any case: Thanks for the quick reply.

  Marc Rohlfing

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[Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06

2006-06-13 Thread Marc Rohlfing
  Hi,

I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) and
Asterisk (to 1.2.9.1) at the same time. Now, when trying to compile
mpg123 - using the tried and true make mpg123 -, the build fails with
an error

make[3]: Entering directory `/usr/src/asterisk-1.2.9.1/mpg123-0.59r'
make[3]: *** No rule to make target `\
', needed by `mpg123'.  Stop.

Maybe there's someone out there more versed in Linux who has an idea
what might have gone wrong. Thanks!

  Marc

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AW: [Asterisk-Users] Free/Open pci telco card

2006-05-23 Thread Marc Rohlfing
  Hi,

 While I was surfing the net last week,
 I found a link for  open source pci telco cards.
 I'm not sure if it were isdn or analog related.
 The layout an all the stuff was free downloadable, so that 
 you can build your own cards.
 Does anybody have the link?

you're probably talking about the Zapata Telephony Project and their 
Tormenta-Cards:
  http://www.zapatatelephony.org/

Marc

-- 
Marc Rohlfing
Training Manager
OASY AG
Heinz-Nixdorf-Ring 1
Gebäude C
33106 Paderborn
 
Telefon:  +49 5251 / 68893 -13
Fax:  +49 5251 / 68893 -09
E-Mail:   [EMAIL PROTECTED]
 
Internet: www.oasy.de
 
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[Asterisk-Users] Asterisk evaluating CLIP, then getting out of the way

2006-04-10 Thread Marc Rohlfing
  Hi,

I've just come upon an interesting question regarding the use of Asterisk as an 
Application Server connected behind a conventional ISDN PBX: The user wants 
to forward all incoming calls through the PBX to Asterisk over S0-Lines, have 
Asterisk do some processing (which includes looking up the final internal 
extension for the incoming call) and then returning the call to the PBX (a 
Siemens HiPath 3000) and have that ring the phone.
The catch: In order not to tie up 2 B-channels for every call on the leg 
between PBX and * (thus needing a big, expensive ISDN card in the Asterisk 
server), we would like to have Asterisk 'drop' the call completely once it has 
been returned to the HiPath PBX.

I've tried to dig up information on the web, but wasn't able to find out 
whether that is possible at all - any feedback is appreciated!

Marc Rohlfing

-- 
Marc Rohlfing
Training Manager
OASY AG
Heinz-Nixdorf-Ring 1
Gebäude C
33106 Paderborn
 
Telefon:  +49 5251 / 68893 -13
Fax:  +49 5251 / 68893 -09
E-Mail:   [EMAIL PROTECTED]
 
Internet: www.oasy.de

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