Hi *,
I have a very simple setup, since this is my first test with asterisk: I have
configured an
Asterisk server and a kphone client (SIP) to talk to each other. Right now, the
SIP user gets
authenticated by asterisk without problems. My goal is to redirect the call to a
given ISDN
telephone number.
Here are the parameters that I want to use for my setup:
- SIP user: test_sip_user
- destination ISDN telephone number: 123456
- asterisk server:
192.168.1.100
ISDN interface
extension associated to the ISDN number: 100
- kphone client:
192.168.1.200
sound system (ALSA with OSS emulation, working)
The first problem that I have is that, even though kphone and asterisk are able
to authenticate
the user, I am not sure that sound gets transmitted.
This is the first thing that I would like to achieve: to verify that sound is
flowing between
kphone and asterisk. The easiest thing would be to get a dial tone in the kphone
client, but I
fear that this is not possible, since SIP initiates a session with all needed
parameters, and does
not need/accept a dial tone. Please, correct me (and tell me how to do it :) )
if I am wrong on
this one.
The next method to verify the flow of sound, easy enough for me to try, would be
to set up a
single mailbox, with a greeting message and the possibility to record speech on
the mailbox. This
should allow me to verify the flow of sound if both directions. Could you
provide any hints on how
to do this? Just a very simple setup is needed.
Once I have verified that sound is flowing, I would like to make the call into
the ISDN network. I
have some questions:
1) Is it actually possible to implement this scenario? I have understood that
asterisk can work as
a gateway between SIP and ISDN (and between other networks, too). Is this
correct?
2) I am not able to figure out what extension to use for the SIP user. The
kphone sends the
following request to asterisk:
sip:100@192.168.1.100:5060
I do not know how to use this in an extension specification in order to get
asterisk to dial the
desired number (123456) via the ISDN interface. I have tried to setup extension
100 to playback a
sound file, like this:
exten => 100,1,Wait(1)
exten => 100,2,Playback(demo-congrats)
exten => 100,3,Hangup
but kphone complains that the session can not be established. What extension
specification should
I use to match the SIP call?
And I have an aside question: kphone can (apparently) also be used for
video-conferences. Is this
in any way supported by asterisk? My impression is that asterisk only provides
voice services.
Thanks for your help,
Daniel Gonzalez