[Asterisk-Users] No Voice Path With PSTN Call Forward

[Philippe Daoust]

Hello list,

I've run into a bizarre problem with a specific situation involving call 
forwarding.

Asterisk is registered to a Nortel MCS5200 system using SIP.  I can make 
outgoing calls and get incoming calls no problem for both internal calls and 
calls to/from the PSTN (the Nortel MCS connects to a bunch of Audiocodes PRI 
GW using SIP for PSTN access).

The issue I encounter is when I call forward my centrex line at the office 
to the DID associated with my SIP account on the MCS5200.  If someone calls 
my office number it gets call forwarded as expected to my DID through the 
PSTN.  The PSTN dials the Audiocodes GW which then signals the MCS5200 and 
finally my * system.  The phones connected to * ring, I answer and get no 
media.  The person calling gets the communication cut off (hangup).  I get 
the following error on the * console:

Nov 17 11:34:16 WARNING[-1092846672]: chan_sip.c:2676 process_sdp: 
Insufficient information for SDP (m = '', c = '')
Nov 17 11:34:16 WARNING[-1092846672]: chan_sip.c:4914 get_rdnis: Huh?  Not 
an RDNIS SIP header ()?

This only happens if I get an incoming call that comes from a forwarded 
line.  Just to be sure, I registered a Sipura SPA-2000 directly to the 
MCS5200 and that works fine.  This leads me to beleive I am missing some 
parameter in my config or that there is some sort of bug in *.

Here is the relevant sip.conf portion (I'm running * 1.0.0) :

[mcs5200]
type=friend
secret=secret
username=username
fromuser=username
fromdomain=whateverdomain.ca
host=mcsaddress.ca
disalow=all
allow=ulaw
insecure=very
dtmfmode=inband

Not sure what to think of this one.  I've done lots of searching but didn't 
come up with anything...

Any help appreciated.

-Regards,