[asterisk-users] Asterisk G722

[rachid]

Hello,

I have some problems to use G722, when my client sent an invite request 
to asterisk using G722/16000 codec
asterisk respond with G722/8000 codec.

I dont know exactly if Asterisk supports G722/16000 codec??
If yes how can I activate It??

Thanks.

Rachid.

Below wireshak trace:

<-------------------------------------------------------------------------------------------------------------------->

My Invite:

INVITE sip:600 at asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
Contact: <sip:Manager at 192.168.10.12:5060>
Authorization: Digest username="Manager", realm="asterisk", 
nonce="1c8c3fd9", uri="sip:600 at asterisk", 
response="5d32f87fa423cd2f1bf9aefb8cf920b6", algorithm=MD5
Max-Forwards: 70
User-Agent: wengo/v1/wengophoneng/wengo/rev54/trunk/
Expires: 120
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length:   365

v=0
o=userX 20000001 20000001 IN IP4 192.168.10.12
s=A call
c=IN IP4 192.168.10.12
t=1202402970 1202406570
m=audio 10600 RTP/AVP 0 8 109 3 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:109 G722/16000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
m=video 10702 RTP/AVP 34 31
a=rtpmap:34 H263/90000/1
a=rtpmap:31 H261/90000/1

<-------------------------------------------------------------------------------------------------------------------->

Asterisk response:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.10.12:5060;branch=z9hG4bK2600322761;received=77.203.231.140
From: Manager <sip:Manager at 91.121.31.80>;tag=3871604470
To: <sip:600 at asterisk>;tag=as5c1447b6
Call-ID: 3325182877 at 192.168.10.12
CSeq: 21 INVITE
User-Agent: Asterisk PBX SVN-trunk-r102777
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:600 at 91.121.31.80:5060>
Content-Type: application/sdp
Content-Length: 397

v=0
o=root 1999706631 1999706631 IN IP4 91.121.31.80
s=Asterisk PBX SVN-trunk-r102777
c=IN IP4 91.121.31.80
b=CT:384
t=0 0
m=audio 18950 RTP/AVP 109 0 8 101
a=rtpmap:109 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18692 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

<-------------------------------------------------------------------------------------------------------------------->