cisco 7821 and codec g722

[Evgeny Andreev]

Hello,

In datashit g722 is supported http://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-ip-phone-7800-series/data-sheet-c78-729488.html  but on device page of 7821 phone i couldn't see option "Advertise G.722".

On phone: Applications - Admin Setting - Status - Call Statistics shows that only G.711 is using.

- Service parameter - g722 codec enabled: enable for all devices except recording-enabled device

- 7821 firmware sip78xx.10-2-1-12

- cucm 8.6.2

Best regards

[inderpreetsingh23]

Hello,

What is the codec setting that you have applied in your region settings? Make sure you have set G722 over there. 

[Aditya Gupta]

Hi 

G722 will only be used if both the endpoints you are using support G722. Then only G722 will  be negotiated.

If you have selected the region settings as 64kbps then by default G722 have the higher changes of being negotiated since it has higher preference over G711. Please ensure G722 is enabled at the following places in CUCM :

1. Enterprise Parameter : Advertise G.722 Codec

2. Service Parameter: g722 codec enabled

3. Cisco Unified Communications Manager Administration--> Device-- > Phone --> Advertise G.722 Codec

Regards

Aditya Gupta

[amralrazzaz]

i have same problem i tried to choose g722-64k but giving me busy tone when im making a call 

im using sip78xx.12-1-1-12 also im using cme for configurations 

=====================
Version 10.5
Max phoneload sccp version 17
Max dspfarm sccp version 18
Cisco Unified Communications Manager Express
For on-line documentation please see:
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/tsd_products_support_series_home.html

protocol mode default
ip source-address 192.168.2.207 port 2000
ip qos dscp:
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for serviceservice directed-pickup
max-ephones 58
max-dn 64
max-conferences 8 gain -6
dspfarm units 0
dspfarm transcode sessions 0
conference software
privacy
no privacy-on-hold
hunt-group report delay 1 hours
hunt-group logout DND
max-redirect 10
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line-mode single
time-format 24
date-format dd-mm-yy
timezone 25 Egypt Standard/Daylight Time
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
no transfer-pattern is configured, transfer is restricted to local SCCP phones only.
keepalive 30 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout transfer-recall 0
timeout ringin-callerid 8
timeout night-service-bell 12
caller-id name-only: enable
web admin system name romani password cisco
web admin customer name romani1 password cisco
edit DN through Web: enabled.
edit TIME through web: enabled.
background save interval 10 minutes
Log (table parameters):
max-size: 150
retain-timer: 15
create cnf-files version-stamp 7960 Jun 03 2018 17:11:36
transfer-system full-consult dss
transfer-digit-collect new-call
local directory service: enabled.
Extension-assigner tag-type ephone-tag.

Router(config-register-pool)#codec ?
g711alaw Use G.711 Alaw
g711ulaw Use G.711 ulaw
g722-64K Use G722-64K 64000 bps
g729r8 Use G.729
ilbc Use iLBC 13330 or 15200 bp

why it giving me busy tone when im trying make call after changing the codec of one phone to g722-64k ????