HI!
I configured asterisk to send all outgoing calls to our Gateway. I noticed
when asterisk sends call to gateway that he represents all calls as
asterisk and not as callerID(number of sjphone client registerd to
asterisk).
Can anyone give me an example of such configuration?
Thank you
HI,
I would like to know how can I check if gateway is registered with gnugk?
Thank you,
Mitja
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Hello!
How can I check if oh323 is loaded and working? Is there a quick test for this?
Thank you.
Micko
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I have a problem when dialing from outside line to sip server. I get this
output on debug.
Could someone give me a hint what could be wrong?
== Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to
exten 's'
== Starting OH323/R30149 at from-pstn,s,1 still failed so falling
I have a problem when dialing from outside line to sip server. I get this
output on debug.
Could someone give me a hint what could be wrong?
== Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to
exten 's'
== Starting OH323/R30149 at from-pstn,s,1 still failed so falling
Hi all
I have a following problem. I want to use sjphone to connect to asterisk sip
server and then I want asterisk to do a conversion to h323 and send this to
h323 gateway.
sjphone---sipASTERISKh323-GATEWAY
Example:
if someone from plane PSTN line dials 123456 the gateway will
a GateKeeper ID from Asterisk to
carriers that authenticate based on that in the past.
Regards,
Sahil Gupta
VoiceValley
On Mon, 16 May 2005, Micko wrote:
Hi all
I have a following problem. I want to use sjphone to connect to asterisk
sip server and then I want asterisk to do
I have a TDM400P with fxo and fxs modules, and local extension calls work fine
but
I can not dial-out. I am getting both problems on FreeBSD and Linux, and
plugging
a regular telefon set works fine. Any ideas of what I am missing?
thanks
micko
Zaptel Configuration
?
thanks
micko
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