Displaying 10 results from an estimated 10 matches for "spielmann".
2005 Jan 07
4
Monitoring
...t happens is:
- if I specify WAV as the format, the resulting files are exactly 44 bytes big
and contain nothing at all
- if I specify GSM as the format, the resulting files are of size 0.
I did not request mixing of the files or anything else.
Any ideas why the monitoring fails?
Cheers
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2005 Jan 18
9
# Transfers.
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not "Enter my PIN followed by Pound"
Likewise if I turn off the ability to transfer when initiating a call,
my bank pin
2005 Jan 10
0
AGI EXEC trouble
...#39;
I do "EXEC ChanIsAvail SIP/phone1", Asterisk says
Jan 10 14:33:20 WARNING[10567]: chan_sip.c:1389 create_addr: No such host:
phone1
What the..? I use Asterisk CVS-v1-0 at the moment.
Oh by the way, some other AGI commands refuse to work:
ANSWER
HANGUP
GET VARIABLE
Cheers
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2005 Jan 13
0
current CVS version
...an_zap.c:9645: (Each undeclared identifier is reported only once
chan_zap.c:9645: for each function it appears in.)
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
I downloaded zaptel, libpri and asterisk. Any hints?
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2005 Jan 13
2
about AGI command parsing
...quot;HANGUP" I get "510 Invalid or unknown
command", if I put something behing "HANGUP" I get return value -1, if I put
a channel name behind "HANGUP", I also get return value -1.
Maybe I'm missing something, or is the AGI command parser broken? :-)
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2005 Jan 15
2
No more loading asterisk...
Hey, whenever I try to load, I get these errors
Jan 15 16:37:24 ERROR[7573]: chan_iax2.c:7486 load_module: Unable to
bind to 0.0.0.0 port 4569: Address already in use
Jan 15 16:37:24 WARNING[7573]: loader.c:345 ast_load_resource:
chan_iax2.so: load_module failed, returning -1
== Manager unregistered action IAXpeers
== Unregistered channel type 'IAX2'
Jan 15 16:37:24 WARNING[7573]:
2005 Jan 17
0
AGI / Sockets
...P/phone1|20|tr)
etc. - if 10.0.0.1 isn't reachable or doesn't react on the connection? In my
test cases, I always got a hangup and no further processing of the dialplan.
Any hints? ( the call mustn't go into Nirvana if the AGI server isn't
available!)
Thanks for any help
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2005 Jan 17
2
Does Asterisk do that?
Hello.
I have just arrived to Asterisk. I would like to know if Asterisk can
perform some functionalities I am looking for.
I want to allow voip over sip to some users. All of them must have
their own user name and password to login to Asterisk so only allowed
users can login. All calls started by users have to be redirected to
one account at our voip provider. I think those functionalities can
2005 Feb 08
12
SRV lookups
...all - for example, there may be john@doe.com and john@bar.com, both get
mapped to my central Asterisk server - I'm unable to know which of the john.s
is being called, hence I cannot route the call correctly.
Hope the question is clear enough ;)
TIA,
Robert
--
Mit freundlichen Gr??en
Robert Spielmann
-----------------------------------------------------
TAL.DE ?Klaus Internet Service GmbH ?spielmann@tal.de
Robertstr. 6 ? ? ? ?* ? ? ?D-42107 Wuppertal, Germany
Tel +49 (0) 202 495-364 ?* ?Fax +49 (0) 202 / 495-399
2006 Dec 04
0
Paypal and soap4r gems
WSDL conversion as proposed in the paypal plugin howto doesn''t work. Any
idea what goes wrong? See the output below.
rsp@hollerith:~/workspace/pptest/vendor/plugins/paypal$ wsdl2ruby.rb
--wsdl http://www.sandbox.paypal.com/wsdl/PayPalSvc.wsdl --type client
--force
F, [2006-12-04T16:30:43.136405 #11579] FATAL -- app: Detected an
exception. Stopping ... undefined method `new'' for