[cisco-voip] CME 12.5 SIP

Sun Aug 25 23:33:48 EDT 2019

This behavior sounds as though the phone thinks it’s registered to the CME, but isn’t..
What are the outputs of these commands?
show voice register pool all brief
show voice register global
show voice register pool <pool tag>

Also, take a packet capture from the CME if you can to see if any SIP packets are making it to the router. It could also be a binding problem on the interface.

Regards
Sreekanth

From: cisco-voip <cisco-voip-bounces at puck.nether.net> On Behalf Of Kent Roberts
Sent: Sunday, August 25, 2019 9:42 AM
To: Heim, Dennis <Dennis.Heim at wwt.com>
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] CME 12.5 SIP

Have you tried it with tcp on that dial peer?   I know it sounds stupid, but never know these days…..

On Aug 24, 2019, at 10:07 PM, Heim, Dennis <Dennis.Heim at wwt.com<mailto:Dennis.Heim at wwt.com>> wrote:

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<http://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
<image001.gif><https://twitter.com/CollabSensei>
<image002.gif><xmpp:dennis.heim at wwt.com><image003.gif><tel:+13142121814><image004.gif><sip:dennis.heim at wwtatc.com>

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