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SET SIP_CODEC and Video issues

Greetings List.
I Have a small test server and i’m facing a small issue.
i have setup two SIP PEERS and they are able to do Video calls.
now I’m testing SET SIP_CODEC in a dial plan and when ever i’m setting the codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW CHANNELS shows only the Codec i set in the dialplan.
is it possible to avoid this problem?

Asterisk version
1.8.11.0

SIP.CONF
=======

[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p

[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p

EXTENSIONS.CONF
[DER-TEST]
;exten => _.,1,NoCDR()
exten => _.,1,Set(SIP_CODEC=alaw)
exten => _.,2,Set(SIP_CODEC_OUTBOUND=gsm)
;exten => _.,2,Set(SIP_CODEC_INBOUND=gsm)
exten => _.,n,DIAL(SIP/TK${EXTEN})
exten => h,1,Hangup()

Tarek Sawah

Information Technology Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993

Best PBX for Call Centers?

Hello List,
I have customer with a 40 Agents call center. and is looking to install a PBX switch that can serve those agents.
As per my experience i suggested Asterisk as i have tested it with Call Centers, however he has been advised not to use it although his provider is using Asterisk to send him calls. He has been advised to use Sippy which they claim is more stable than Asterisk.
i’m not an expert with Sippy so i’m looking for a piece of an advise here.. if i’m doing an Asterisk Vs Sippy comparison. can anyone help?
Regards

Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993

USA Did required

Google is your best friend when looking for this type of assistance my friend.
try callcentric vonage packet8 for reliable retail DIDs.

Tarek Sawah

Information Technology Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993

Date: Sat, 1 Oct 2011 00:51:59 +0530
From: amit.magnate@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] USA Did required

Hello members,
I am looking for USA incoming DID which can be registered on softphone/IP Phone/ Pap2 devices.

The DID will only be required to receive inbound calls and no outbound calls.
Let me know your best per month prices/cost for the above.
Regards,
Amit Mehta