Hi

When I make a call from pstn via ser to hit asterisk, then to a local (local to asterisk) sip phone the call goes through, but I can hear the pstn user on the sip phone, but not the other way.

When the pstn is dialed from the sip phone, the call goes via asterisk direct to pstn gw, bypassing ser, in this case two way calling works fine.

I send the call from ser to asterisk using rewritehost, but it seems that voice path back from asterisk to ser to gw is broken. My sequence of events is below...

gw ---> ser INVITE ser ----> gw TRYING ast ----> xlite INVITE xlite ----> ast TRYING xlite ----> ast RINGING ser -----> gw RINGING xlite ------> ast OK ast ------>xlite ACK ast ----> xlite INVITE ser ----> gw OK gw -----> ser ACK ser -----> gw ACK xlite ---> ast TRYING xlite ----> ast OK ast ----> xlite ACK ast -----> xlite INVITE xlite ---> ast TRYING xlite --> ast OK ast ---> xlite ACK ast ---- xlite INVITE xlite ---> ast TRYING xlite ---> ast OK ast ---> xlite ACK

Iqbal