[Asterisk-Users] DTMF detection on SIP provider ?
Mikael Andersson
micke at party.pp.se
Sun Mar 9 17:33:36 MST 2003
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote:
>Look into sip.conf.sample
>
>[general]
>port = 5060 ; Port to bind to
>bindaddr = 0.0.0.0 ; Address to bind to
>context = default ; Default for incoming calls
>;tos=lowdelay
>;tos=184
>;maxexpirey=3600 ; Max length of incoming registration we
>allow
>;defaultexpirey=120 ; Default length of incoming/outoing
>registratio
>;
>;register => 1234 at mysipprovider.com ; Register with a SIP provider
>;register => 2345 at mysipprovider.com/1234 ; Register 2345 at sip provider as
>1234
>;
>;[snomsip]
>;type=friend
>;secret=blah
>;host=dynamic
>;dtmfmode=inband <<here is the answer ; Choices are inband,
>rfc2833, or info
>;defaultip=192.168.0.59
Well.. But I need it on :
the [general] part where I do the register ? or ?
The "clients" in my case all my ATAs work fine.. But incoming calls doesnt..
/Mike
More information about the asterisk-users
mailing list