Hello, please help me to choose corret codec in trunk group
I set network region in signalling group,for which i set codec set to choose only G711A G711MU codecs. (attachment)
But when i make a call, there is G729 codec, when i debug a call in router
Where else i set choosing necessary codec for trunk group?
To: <sip:92584040@10.2.0.5>
Date: Tue, 15 Nov 2022 10:13:11 GMT
Call-ID: 1a889b00-1ee182d1-b1d0ae-25e6020a@10.2.230.37
Supported: timer,resource-priority,replaces
Min-SE: 500
User-Agent: Avaya CM/R015x.02.1.016.4
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Session-ID: 5138e0179ab6358156ee96ab96270729;remote=0000000000000000000000000000 0000
Cisco-Guid: 0445160192-0000065536-0000240315-0635830794
Session-Expires: 1800
P-Asserted-Identity: "Pavel Velichko" <sip:20043@kazminerals.com>
Remote-Party-ID: "Pavel Velichko" <sip:20043@kazminerals.com>;party=calling;scre en=yes;privacy=off
Contact: <sip:20043@10.2.230.37:5060>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 277
v=0
o=CiscoSystemsCCM-SIP 96270730 1 IN IP4 10.2.230.37
s=SIP Call
c=IN IP4 10.16.238.252
b=TIAS:48000
b=AS:64
t=0 0
m=audio 10038 RTP/AVP 8 0 18 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15
I set network region in signalling group,for which i set codec set to choose only G711A G711MU codecs. (attachment)
But when i make a call, there is G729 codec, when i debug a call in router
Where else i set choosing necessary codec for trunk group?
To: <sip:92584040@10.2.0.5>
Date: Tue, 15 Nov 2022 10:13:11 GMT
Call-ID: 1a889b00-1ee182d1-b1d0ae-25e6020a@10.2.230.37
Supported: timer,resource-priority,replaces
Min-SE: 500
User-Agent: Avaya CM/R015x.02.1.016.4
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=MIXED
Session-ID: 5138e0179ab6358156ee96ab96270729;remote=0000000000000000000000000000 0000
Cisco-Guid: 0445160192-0000065536-0000240315-0635830794
Session-Expires: 1800
P-Asserted-Identity: "Pavel Velichko" <sip:20043@kazminerals.com>
Remote-Party-ID: "Pavel Velichko" <sip:20043@kazminerals.com>;party=calling;scre en=yes;privacy=off
Contact: <sip:20043@10.2.230.37:5060>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 277
v=0
o=CiscoSystemsCCM-SIP 96270730 1 IN IP4 10.2.230.37
s=SIP Call
c=IN IP4 10.16.238.252
b=TIAS:48000
b=AS:64
t=0 0
m=audio 10038 RTP/AVP 8 0 18 127
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-15