[Asterisk-Users] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio

[Michael J. Tubby B.Sc (Hons) G8TIC]

Gents,

I've recently built a couple of Asterisk boxes and want to migrate
away from CallManager to Asterisk.

On my Asterisk box I have about 8 Grandstream BT101s and a
Cisco 7905G in SIP mode, on my CallManager I have about 10
x 30VIP, 2 x 7940 and a 7960.

I've built Asterisk version 1.0.5 along with Zozo's chan_sccp
(CVS latest from last night) and got it partially working.  All devices
are on the inside of a private network at the moment (192.168.144.0/24)
and I'm having some issues with devices on chan_sccp.

The 30VIPs can place and receive calls but I have a one-way
audio problem.  The 7960 can receive calls but when I place calls
from it I end up directly in the voicemail "unavailable" and the SIP
phone doesn't ring.

Looking at the network the SIP device opens an RTP stream to the
Cisco (30VIP or 7960) but the Cisco device doesn't send RTP
back to the SIP phone...  can anyone point me in the right direction
with this?

A more general question: with Cisco phones being removed from
a CallManager environment, is it best to keep them in Skinny/SCCP
mode or change out to SIP?  The 30VIPs can only do SCCP/Skinny
so which of the two channel drivers in Asterisk should I use for
best effect?


Mike