[cisco-voip] CME 12.5 SIP

Sun Aug 25 00:07:42 EDT 2019

I am configuring CME 12.5 with SIP phones. However, when I dial from the phone, I see nothing on the router from either dial-peer or ccsip traces. When I dial any of the numbers such as +13145552002, the SIP phone just sits there and eventually gives you reorder.

ip dhcp pool PSTN-Voice
network 10.1.100.0 255.255.255.0
default-router 10.1.100.1
 option 150 ip 10.1.100.1
 dns-server 10.1.5.14
 domain-name ciscoclass.com
class PSTN-Voice-Range-Class
  address range 10.1.100.100 10.1.100.199

voice service voip
no ip address trusted authenticate
address-hiding
media disable-detailed-stats
allow-connections sip to sip
no supplementary-service sip handle-replaces
sip
  bind control source-interface GigabitEthernet2
  bind media source-interface GigabitEthernet2
  registrar server expires max 1200 min 300
  early-offer forced
!
voice class codec 1
codec preference 1 g711ulaw

voice register global
mode cme
source-address 10.1.100.1 port 5060
max-dn 10
max-pool 10
authenticate register
authenticate realm all
timezone 8
service https
tftp-path flash:
create profile sync 0274021104025313
auto-register
!
voice register dn  1
number +13175551000
name PSTN Phone
label PSTN-3175551000
!
voice register pool  1
id mac 3CCE.7359.9A93
type 7942
number 1 dn 1
dtmf-relay rtp-nte
username <user> password <password>
codec g711ulaw
no vad
!
interface GigabitEthernet1
description ** SIP SP **
ip address 10.1.20.1 255.255.255.0
ip tcp adjust-mss 1360
negotiation auto
no mop enabled
no mop sysid
!
interface GigabitEthernet2
description ** PSTN Voice **
ip address 10.1.100.1 255.255.255.0
ip tcp adjust-mss 1360
negotiation auto
no mop enabled
no mop sysid
!
interface GigabitEthernet3
description ** Management **
ip address 192.168.15.52 255.255.255.0
negotiation auto
no mop enabled
no mop sysid
!
ip route 0.0.0.0 0.0.0.0 192.168.15.1
!
tftp-server bootflash:apps42.9-4-2ES26.sbn
tftp-server bootflash:cnu42.9-4-2ES26.sbn
tftp-server bootflash:cvm42sip.9-4-2ES26.sbn
tftp-server bootflash:dsp42.9-4-2ES26.sbn
tftp-server bootflash:jar42sip.9-4-2ES26.sbn
tftp-server bootflash:SIP42.9-4-2SR3-1S.loads
tftp-server bootflash:term42.default.loads
tftp-server bootflash:term62.default.loads
!
elephony-service
load 7945 term42.default
max-conferences 8 gain -6
transfer-system full-consult
!
!
dial-peer voice 100 voip
description *** Inbound LAN side dial-peer ***
session protocol sipv2
incoming called-number .
voice-class codec 1
 voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
!
dial-peer voice 500 voip
description ** HQ DIDs **
destination-pattern +1314555200[0-9]
session protocol sipv2
session target ipv4:10.1.20.10
session transport udp
voice-class codec 1
 voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
!
dial-peer voice 501 voip
description ** BR DIDs **
destination-pattern +1206555300[0-9]
session protocol sipv2
session target ipv4:10.1.20.20
session transport udp
voice-class codec 1
 voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
!
Dennis Heim | Domain Architect (Collaboration Labs)
World Wide Technology, Inc. | +1 314-212-1814
[cid:image001.png at 01D10DD2.7FC81F90]<https://twitter.com/CollabSensei>
[cid:image002.png at 01D10DD2.7FC81F90]<xmpp:dennis.heim at wwt.com>[cid:image003.png at 01D10DD2.7FC81F90]<tel:+13142121814>[cid:image004.png at 01D10DD2.7FC81F90]<sip:dennis.heim at wwtatc.com>

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