02-26-2014 12:05 AM - edited 03-18-2019 02:39 AM
Hello everybody!
I need help so much!
I have CME (IOS 15.1(4)M2 and CME 8.6) and CUCM 7.1
I have third-party SIP client on CME - and 9951 on CUCM.
I try to make a video call. but still cannot.
every call maiden is audio only.
inside each site - inside CUCM or inside CME - everything is working fine with video
here is a part of my CME config:
voice service voip
no ip address trusted authenticate
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol none
modem passthrough nse codec g711ulaw
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server
asymmetric payload full
pass-thru content sdp
voice class codec 4101
codec preference 1 g711ulaw
codec preference 2 g711alaw
video codec h261
video codec h263
video codec h263+
video codec h264
video codec mpeg4
voice register global
mode cme
source-address 192.168.AB.CDE port 5060
max-dn 30
max-pool 30
authenticate realm all
timezone 32
tftp-path flash:
file text
create profile sync 031109583450814A
camera
video
voice register dn 2
number 188
voice register pool 2
id mac 0000.0000.0000
number 2 dn 2
dtmf-relay rtp-nte
username 188 password 188188
codec g711ulaw
no vad
camera
video
sccp local GigabitEthernet0/1
sccp ccm 192.168.AB.CDE identifier 1 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/1
associate ccm 1 priority 1
associate profile 1 register XCODE
keepalive retries 5
switchover method immediate
switchback method immediate
switchback interval 15
dspfarm profile 1 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
codec g729r8
maximum sessions 4
associate application SCCP
dial-peer voice 201 voip
translation-profile outgoing TO_OFFICE
destination-pattern 700...
session protocol sipv2
session target ipv4:192.168.KL.MNO
voice-class codec 4101
no voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
no vad
telephony-service
sdspfarm units 5
sdspfarm transcode sessions 10
sdspfarm tag 1 XCODE
no auto-reg-ephone
max-ephones 30
max-dn 30
ip source-address 192.168.AB.CDE port 2000
max-conferences 8 gain -6
transfer-system full-blind
transfer-pattern .T
create cnf-files version-stamp 7960 Feb 24 2014 12:00:01
on CUCM I have SIP trunk with DTMF - RFS 2833
So where am I wrong? please help!!!
03-02-2014 10:37 PM
anybody? please help!
05-31-2014 09:59 AM
A bit late, but try this:
dial-peer voice 201 voip
voice-class sip pass-thru content sdp
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide