Philip Prindeville wrote:
I did a little investigation into my one-way voice issue, and noticed
that if I don't do voice-menus (i.e. where the Asterisk box itself
generates the first outbound INVITE, then passes-through the 2nd INVITE
once a handset picks up) then I get two-way voice (i.e. with sending the
call directly to the phone). (In this topology, my Asterisk box is also
my firewall/NATting router...)
If I enable the voice menus in the inbound dialplan, however, it can
hear the voice menus, but not the called-party when they pick up their
phone (extension).
So someone (either the SIP conntrack module on the Asterisk border
firewall or else the SBC at the ILEC) is failing to look into the 2nd
INVITE (i.e. we're not rewriting it properly as it goes by, or the SBC
is failing to see it).
What module options are you using for the SIP helper and how is call
setup in asterisk configured (directrtpsetup, canreinvite, ...)?
--
To unsubscribe from this list: send the line "unsubscribe netfilter-devel" in
the body of a message to majordomo@xxxxxxxxxxxxxxx
More majordomo info at http://vger.kernel.org/majordomo-info.html
