Cant get h264 working with polycom vvx1500

I am working with Asterisk 1.6.2.10. I have someone calling to me with a Tandberg sending h264.
Audio is working fine, but I cannot get video working. Any and all help would be appreciated (by my wife and kids as well, I’ve been working on this non-stop for 17 hours). Thanks!!!

I’ve added this to sip.conf
[general]
videosupport=yes
allow=h264

and this to sip_additional.conf (500 is the extension for the polycom)
[500]
deny=0.0.0.0/0.0.0.0
secret=1234
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
allow=h264
dial=SIP/500
mailbox=500@default
permit=0.0.0.0/0.0.0.0
callerid=device <500>
callcounter=yes
faxdetect=no

[inbound]
host=proxy-east.xxxxxx.com
username=xxxxxxxxxx
secret=xxxxxxxxxx
type=friend
dtmfmode=auto
context=from-trunk ; (this could be ext-did or from-pstn as well)
insecure=port,invite
canreinvite=no
nat=yes
allow=h264
allow=ulaw

I can see the incoming guy trying to pass me video:
m=video 32816 RTP/AVP 97 98.
b=TIAS:768000.
a=rtpmap:97 H264/90000.
a=fmtp:97 profile-level-id=42800d;max-mbps=40500;max-fs=1344;max-smbps=40500.
a=rtpmap:98 H264/90000.
a=fmtp:

But I’m returning video 0:
s=Asterisk PBX 1.6.2.10.
c=IN IP4 xxx.xxx.xxx.117.
t=0 0.
m=audio 30584 RTP/AVP 0 102.
a=rtpmap:0 PCMU/8000.
a=rtpmap:102 telephone-event/8000.
a=fmtp:102 0-16.
a=ptime:20.
a=sendrecv.
m=video 0 RTP/AVP 97 98.

Other video related items in the logs:
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing session-level SDP v=0… UNSUPPORTED.

[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found RTP audio format 0
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found RTP audio format 101
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing media-level (audio) SDP a=sendrecv… OK.
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found audio description format PCMU for ID 0
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000… OK.
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found audio description format telephone-event for ID 101
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found RTP video format 99
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing media-level (video) SDP a=sendrecv… OK.
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Found video description format H264 for ID 99
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: Processing media-level (video) SDP a=rtpmap:99 H264/90000… OK.
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x4 (ulaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x200004 (ulaw|h264)
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Peer audio RTP is at port 192.168.1.103:2322
[Oct 1 02:25:24] VERBOSE[2868] chan_sip.c: Peer doesn’t provide video
[Oct 1 02:25:24] DEBUG[2868] chan_sip.c: We’re settling with these formats: 0x200004 (ulaw|h264)

[Oct 1 02:25:24] VERBOSE[6023] app_dial.c: – SIP/500-00000022 answered SIP/xxxxxx.com-00000021
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: SIP answering channel: SIP/xxxxxx.com-00000021
[Oct 1 02:25:24] DEBUG[6023] rtp.c: Setting the marker bit due to a source update
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: Setting framing from config on incoming call
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: False Text flag: True
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 1 02:25:24] VERBOSE[6023] chan_sip.c: Audio is at 192.168.1.102 port 10062
[Oct 1 02:25:24] VERBOSE[6023] chan_sip.c: Adding codec 0x4 (ulaw) to SDP
[Oct 1 02:25:24] VERBOSE[6023] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: – Done with adding codecs to SDP
[Oct 1 02:25:24] DEBUG[6023] chan_sip.c: Done building SDP. Settling with this capability: 0x4 (ulaw)
[Oct 1 02:25:24] VERBOSE[6023] chan_sip.c:

Hi have a look at

blog.voipsupply.com/update-on-polycom-vvx-1500

and

viewtopic.php?f=13&t=68138

and

fonality.com/trixbox/node/40785

thanks for the info, looks like the answer is upgrade the firmware on the phone which i cant do because it’s a beta unit and wont accept production bootroms for some silly reason. i’ll give mirial a try instead. thanks for the help, much appreciated.