Rob a écrit :
Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed
[...]
Did you try to call Gizmo numbers to see if you have success with them?
** Hear your Gizmo5 number repeated back to you.
*0 Test your router's SIP compatibility.
411 The
Rob a écrit :
Hi all,
Hi
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
while and it works fine I just added CALL OUT ... I have no problem
with call setup ... the called party hears me ... but I can't hear them
again if the call comes INTO the server
Yes ... as a matter of fact here is the sip.conf ... obviously private info
removed
[general]
register =
1747xxx:x...@proxy01.sipphone.com1747xxx%3ax...@proxy01.sipphone.com
port = 5060
bindaddr = 192.168.22.5
context = incoming
svrlookup=yes
;dtmfmode=inband
I do not use Gizmo for inbound, only out. I have a register line that
looks like yours. In addition I have this:
[general]
context=nonesaid
allowguest=no
allowoverlap=yes
allowtransfer=yes
realm=my system's host name
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
maxexpiry=3600
minexpiry=60
Hi all,
I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a
while and it works fine I just added CALL OUT ... I have no problem
with call setup ... the called party hears me ... but I can't hear them
again if the call comes INTO the server both sides work fine.
Just
