[Serusers] pstn --ser--asterisk ---sip phone

[Iqbal]

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