08-14-2012 03:59 PM - edited 03-16-2019 12:43 PM
Hey there,
Setting: 2 phones (7962's) connected to POE switch (3750) which has a link to 2911 running SIP CME 9.0...
State: Both phones register perfectly with newest SIP code.
Condition: 1 out of every 3 call's from either phone to the other fails... no ring on the destination phone... So one call works... call completes, 2 way talk path, then hang up and redial, and no ringy... Is this potentually a DTMF relay issue? I have dtmf-relay sip-notify enabled on all pools...
Ideas?
See config below:
version 15.1
service timestamps debug datetime msec
service timestamps log datetime
service password-encryption
!
hostname TEST
!
boot-start-marker
boot-end-marker
!
!
logging buffered 20000
enable secret XXXXX
!
no aaa new-model
!
!
no ipv6 cef
ip source-route
ip cef
!
!
!
ip multicast-routing
ip dhcp excluded-address 192.168.222.0 192.168.222.10
!
ip dhcp pool phones
network 192.168.222.0 255.255.255.0
domain-name Test123.com
default-router 192.168.222.1
option 150 ip 192.168.100.1
!
!
no ip domain lookup
ip domain name Test123.com
!
multilink bundle-name authenticated
!
!
!
!
!
!
template 1
!
!
voice-card 0
!
!
!
voice service voip
allow-connections sip to sip
no supplementary-service sip handle-replaces
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
voice register global
mode cme
source-address 192.168.100.1 port 5060
timeouts interdigit 3
max-dn 100
max-pool 50
load 7962 SIP42.9-3-1-1S
authenticate register
authenticate realm all
timezone 6
tftp-path flash:
create profile sync 0006074543342742
ntp-server 192.168.100.1 mode directedbroadcast
!
voice register dn 1
number 1111
name IP Phone1 1111
label IP Phone1 1111
!
voice register dn 2
number 1112
name IP Phone2 1112
label IP Phone2 1112
!
voice register dn 3
number 1113
name IP Phone3 1113
label IP Phone3 1113
!
voice register dn 4
number 1114
name IP Phone4 1114
label IP Phone4 1114
!
voice register dn 5
number 1115
name IP Phone5 1115
label IP Phone5 1115
!
voice register dn 6
number 1116
name IP Phone6 1116
label IP Phone6 1116
!
voice register template 1
button-layout 1-3 line
button-layout 4-6 speed-dial
network-locale 1
user-locale 1
!
voice register pool 1
fastdial 2 1112
fastdial 6 1116
id mac A418.7529.4925
type 7962
number 1 dn 1
template 1
dtmf-relay sip-notify
voice-class codec 1
username test password test
description Phone One
no vad
!
voice register pool 2
id mac A418.7528.5775
type 7962
number 1 dn 2
template 1
dtmf-relay sip-notify
voice-class codec 1
username test2 password test
description Phone Two
no vad
!
voice register pool 3
id mac A418.7529.34DE
type 7962
number 1 dn 3
template 1
dtmf-relay sip-notify
voice-class codec 1
username test3 password test
description Phone Three
no vad
!
voice register pool 4
id mac A418.7528.F2F4
type 7962
number 1 dn 4
template 1
dtmf-relay sip-notify
voice-class codec 1
username test4 password test
description Phone Four
no vad
!
voice register pool 5
id mac A418.7528.F451
type 7962
number 1 dn 5
template 1
dtmf-relay sip-notify
voice-class codec 1
username test5 password test
description Phone Five
no vad
!
voice register pool 6
id mac F4EA.6749.6C1A
type 7962
number 1 dn 6
template 1
dtmf-relay sip-notify
voice-class codec 1
username test6 password test
description Phone Six
no vad
!
!
!
!
license udi pid CISCO2911/K9 sn xxxxx
license accept end user agreement
!
!
username test privilege 15 password 7 XXXXX
!
redundancy
!
!
ip tftp source-interface Loopback0
!
!
interface Loopback0
ip address 192.168.100.1 255.255.255.255
ip ospf 1 area 0
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/2.100
description To_3750
ip address 192.168.222.1 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/0/0
no ip address
!
interface GigabitEthernet0/0/1
no ip address
!
interface GigabitEthernet0/0/2
no ip address
!
interface GigabitEthernet0/0/3
no ip address
!
!
router ospf 1
network 192.168.100.0 0.0.0.255 area 0
!
ip forward-protocol nd
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
!
!
!
!
!
!
tftp-server flash0:/Cisco7962/apps42.9-3-1TH2-5.sbn alias apps42.9-3-1TH2-5.sbn
tftp-server flash0:/Cisco7962/cnu42.9-3-1TH2-5.sbn alias cnu42.9-3-1TH2-5.sbn
tftp-server flash0:/Cisco7962/cvm42sip.9-3-1TH2-5.sbn alias cvm42sip.9-3-1TH2-5.
sbn
tftp-server flash0:/Cisco7962/dsp42.9-3-1TH2-5.sbn alias dsp42.9-3-1TH2-5.sbn
tftp-server flash0:/Cisco7962/jar42sip.9-3-1TH2-5.sbn alias jar42sip.9-3-1TH2-5.
sbn
tftp-server flash0:/Cisco7962/SIP42.9-3-1-1S.loads alias SIP42.9-3-1-1S.loads
tftp-server flash0:/Cisco7962/term62.default.loads alias term62.default.loads
!
control-plane
!
!
!
!
mgcp profile default
!
!
!
!
!
gatekeeper
shutdown
!
!
!
line con 0
login local
line aux 0
line vty 0 4
privilege level 15
login local
transport input ssh
line vty 5 15
privilege level 15
login local
transport input ssh
!
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 1
end
08-14-2012 10:36 PM
For best results, use phones with SCCP firmware, not SIP.
You will have more features, and less bugs.
08-15-2012 08:44 AM
I appreciate your response Paolo. I really would like to get this up and running in SIP CME per the potential of running video phones (9971, etc...) which only run SIP code...
Any good debugging commands that may shed some light on this situation? Here are some that I have tried:
debug ccsip message
debug voip dialpeer all
and multiple others... with the ccsip debug command when the phone doesn't ring, there is no SIP debug dialog logged...
Thanks!
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