Skinny phones = POTS dial peers with voice ports!!
SIP phones = voip dial peers (no voice port created by the SIP phone)
To configure SIP Phones on CME:
Voice register global
mode cme ( by default it is SRST mode, to put it back: “no mode cme”)
source-address <CME IP address> port 5060
max-dn 2
max-pool 2 ( = max-ephone )
tftp-path flash: (the location where the config files will be stored, we can choose a subfolder ..)
create profile ( = create cnf-files BUT we should put it after any change to create config file!! Not a 1 time command)
Voice register dn 1 (= ephone-dn 1)
number 3005
Voice register pool 1 (= ephone 1)
id mac 001B.D4C6.1AA1
type 7960
number 1 dn 1 (= button 1:1)
dtmf-relay rtp-nte
description 3214-3005 (label to show on the phone top)
codec g711ulaw
See config files:
Show voice register tftp-binding ( show SIP phones config files) SIPXXXXXXXXXXXX.cnf
Show telephony-service tftp-binding
( show Skinny phones config files) SEPXXXXXXXXXXXX.cnf.xml
!!Without these files the phone will not be able to register
To debug SIP Phone registration:
debug tftp event
debug ccsip mess
Check to see these messages:
1. “REGISTER sip:10.10.202.1 SIP/2.0” message
2. SIP/2.0 100 Trying
3. SIP/2.0 200 OK
!if a SIP phone attempt to register, it will not register with “Voice register global”, we need to setup the voice registrar server:
Voice service voip
registrar server expire max 600 min 60 ( the registration will expire in 600 minutes max and 60 min max)
!!Unlike SCCP Phones, if we have a dn on the phone this does not mean that the phone is registered with CME!! Because SIP is a “peer to peer” protocol.
Without the registration to CME (if we don’t configure registrar server), we will get dial tone and we will be able to dial another phone BUT inbound calls will not be possible. The CME will not route calls to the phone!!
To test if a SIP phone is registered:
The registration will be done when a 100 trying then a 200 ok message comes in.
Show voice register all
show voice register dial-peers ( see the voip dial peer built point to the IP address of the phone)
show voice register pool 1 (see phone + created dial peer)
+ test an inbound call to that phone.
Dial peer for phone:
Dial-peer voice 40001 voip
destination-pattern 3005
session target
ipv4 :10.10.202.51 :5060
session protocol sipv2
dtmf-relay rtp-nte
codec g711ulaw byte 160
SIP digest authentication mechanism:
This is required if you have SIP phones on different segments!!
We can’t use ID MAC for phones if they are not connected directly to the CME, because the registration request will come with a different Mac than the phone.
So the other option will be to use the SIP digest authentication mechanism:
Ex using xlite client
Voice register global
authenticate register ( this cmd will broke registration of other local phones)
Voice register dn 2
number 3006
Voice register pool 2
id mac 1234.4567.6789
type 7960
number 1 dn 2
dtmf-relay rtp-nte
description 32-14-3006
codec g711ulaw
username phone1 password cisco
Voice register global
create profile ( to create cnf file)
Show tftp-binding
More flash:/SIP12345676789.cnf (see username password)
Configure username / password on the SIP Phone
Authentication user name: phone1
Password: cisco
user name : 3006
Domain> 10.10.202.1 (CME IP)
SIP IP Phones over WAN:
By default the phone will only trust the packets from IP address configured in:
Voice register global
source-address 10.10.202.1
port 5060
So if messages come from the serial interface of the CME with a different IP the phone will not be able to register.
To resolve this issue we should bind SIPmessages:
Voice service voip
SIP
bind all source-interface vlan400 (voice vlan)
SIP phones = voip dial peers (no voice port created by the SIP phone)
To configure SIP Phones on CME:
Voice register global
mode cme ( by default it is SRST mode, to put it back: “no mode cme”)
source-address <CME IP address> port 5060
max-dn 2
max-pool 2 ( = max-ephone )
tftp-path flash: (the location where the config files will be stored, we can choose a subfolder ..)
create profile ( = create cnf-files BUT we should put it after any change to create config file!! Not a 1 time command)
Voice register dn 1 (= ephone-dn 1)
number 3005
Voice register pool 1 (= ephone 1)
id mac 001B.D4C6.1AA1
type 7960
number 1 dn 1 (= button 1:1)
dtmf-relay rtp-nte
description 3214-3005 (label to show on the phone top)
codec g711ulaw
See config files:
Show voice register tftp-binding ( show SIP phones config files) SIPXXXXXXXXXXXX.cnf
Show telephony-service tftp-binding
( show Skinny phones config files) SEPXXXXXXXXXXXX.cnf.xml
!!Without these files the phone will not be able to register
To debug SIP Phone registration:
debug tftp event
debug ccsip mess
Check to see these messages:
1. “REGISTER sip:10.10.202.1 SIP/2.0” message
2. SIP/2.0 100 Trying
3. SIP/2.0 200 OK
!if a SIP phone attempt to register, it will not register with “Voice register global”, we need to setup the voice registrar server:
Voice service voip
registrar server expire max 600 min 60 ( the registration will expire in 600 minutes max and 60 min max)
!!Unlike SCCP Phones, if we have a dn on the phone this does not mean that the phone is registered with CME!! Because SIP is a “peer to peer” protocol.
Without the registration to CME (if we don’t configure registrar server), we will get dial tone and we will be able to dial another phone BUT inbound calls will not be possible. The CME will not route calls to the phone!!
To test if a SIP phone is registered:
The registration will be done when a 100 trying then a 200 ok message comes in.
Show voice register all
show voice register dial-peers ( see the voip dial peer built point to the IP address of the phone)
show voice register pool 1 (see phone + created dial peer)
+ test an inbound call to that phone.
Dial peer for phone:
Dial-peer voice 40001 voip
destination-pattern 3005
session target
ipv4 :10.10.202.51 :5060
session protocol sipv2
dtmf-relay rtp-nte
codec g711ulaw byte 160
SIP digest authentication mechanism:
This is required if you have SIP phones on different segments!!
We can’t use ID MAC for phones if they are not connected directly to the CME, because the registration request will come with a different Mac than the phone.
So the other option will be to use the SIP digest authentication mechanism:
Ex using xlite client
Voice register global
authenticate register ( this cmd will broke registration of other local phones)
Voice register dn 2
number 3006
Voice register pool 2
id mac 1234.4567.6789
type 7960
number 1 dn 2
dtmf-relay rtp-nte
description 32-14-3006
codec g711ulaw
username phone1 password cisco
Voice register global
create profile ( to create cnf file)
Show tftp-binding
More flash:/SIP12345676789.cnf (see username password)
Configure username / password on the SIP Phone
Authentication user name: phone1
Password: cisco
user name : 3006
Domain> 10.10.202.1 (CME IP)
SIP IP Phones over WAN:
By default the phone will only trust the packets from IP address configured in:
Voice register global
source-address 10.10.202.1
port 5060
So if messages come from the serial interface of the CME with a different IP the phone will not be able to register.
To resolve this issue we should bind SIPmessages:
Voice service voip
SIP
bind all source-interface vlan400 (voice vlan)
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