I've received a polish did on www.newfon.pl
I've successfully called myself using the sipsettings & voipcheap to perform a free call between my fixed line & the free did... quality was ok
I've copied those settings in my pbxes trunk...
the trunk goes green in status...
but my call doesn't seem to end on pbxes (RINGING....) and nothing in the Call Log of pbxes?
outbound proxy : dialnet.pl
user : the DID number
password: ...
nothing special ?
when i configure Xten lite behind my natted router... the call comes through.... is pbxes in some kind of blacklist?
Did you have to use the Xten Lite they provide in their site, or where you able to provision the account credentials on another soft-client or ATA successfully?
That's correct...
I tried sjphone, eyebeam, xlite 3, xlite 2
their client is an xlite 2 version;
So i've tried to replicate their settings in xlite 2...
there was only 1 variable I couldn't replicate which was
Use X-Nat to choose SIP/RTP Ports: Default
in xlite you can not change this variable and this is "NEVER"...
According to their support pages it should work on any type of ATA adaptor.. but they didn't want to give any asterisk assurance...
