External SIP Trunk Registration not staying up- Drops after every call

rka61 · ‎01-06-2013

Hi All

I am hoping you can help me with a problem that I am facing with sip trunk registration. I am newbiee to setting up CCME so please bear with me

Scenario: Have a Cisco 1861 router running c1861-spservicesk9-mz.152-2.T1.bin on which i have installed CCME

The router is also acting as my home ADSL router. I have 2 hard phones plugged into it and both phones register and I can all each other internally no problems. Both Cisco phone are running SCCP.

I have setup a SIP trunk with voipdiscount.com and when I reboot the router the SIP trunk come up automically and I have configured CCME for outbound call only at this time and it all works fine.

Version of CCME

Version 9.0

Max phoneload sccp version 17

Max dspfarm sccp version 18

Cisco Unified Communications Manager Express

Problem: After about an hour or so of no activity the SIP registration drops and when I try to make any outbound calls I just get the engaged tone. When I run this command it show the registration as down.

Cisco1861#show sip-ua register st

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

5001 20001 119 no

5002 20002 119 no

5003 20003 120 no

rka61xxx -1 119 no

rka61xxx is the SIP trunk

Workaround: Only way to bring the SIP trunk back up all I have to do is ping the sip trunk I am using

Cisco1861#ping sip.voipdiscount.com

Translating "sip.voipdiscount.com"...domain server (8.8.8.8) [OK]

Type escape sequence to abort.

Sending 5, 100-byte ICMP Echos to 77.72.169.131, timeout is 2 seconds:

!!!!!

Success rate is 100 percent (5/5), round-trip min/avg/max = 48/49/52 ms

Cisco1861#ping sip.voipdiscount.com

Type escape sequence to abort.

Sending 5, 100-byte ICMP Echos to 77.72.169.131, timeout is 2 seconds:

!!!!!

Success rate is 100 percent (5/5), round-trip min/avg/max = 52/52/52 ms

Then if I wait a minute and then run this command the SIP trunk is back up

Cisco1861#show sip-ua register st

Line peer expires(sec) registered P-Associ-URI

================================ ========== ============ ========== ============

5001 20001 2847 yes

5002 20002 2846 yes

5003 20003 2847 yes

rka61xxx -1 2846 yes

Now I can make outbound calls. But if I make a call and hangup and make another call I get the engaged tone and the SIP trunk goes down again.

I do not know why this is the case and have to ping it again to bring it back up as you can imagine this is very frustrating. I have contacted my SIP provider and they do not state the problem is at there end. These are the SIP details to use http://www.voipdiscount.com/sip

I am attaching my config

version 15.2

service timestamps debug datetime msec

service timestamps log datetime msec

service password-encryption

!

hostname Cisco1861

!

boot-start-marker

boot-end-marker

!

no aaa new-model

mmi polling-interval 60

no mmi auto-configure

no mmi pvc

mmi snmp-timeout 18

dot11 syslog

!

no ip dhcp use vrf connected

ip dhcp excluded-address 10.0.0.41 10.255.255.254

!

ip dhcp pool LAN

import all

network 10.0.0.0 255.255.255.0

update dns

dns-server 8.8.8.8 8.8.4.4

default-router 10.0.0.253

option 150 ip 10.0.0.253

lease 10

!

ip name-server 8.8.8.8

ip name-server 8.8.4.4

ip cef

no ipv6 cef

multilink bundle-name authenticated

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

sip

asserted-id pai

localhost dns:home.org

outbound-proxy dns:sip.voipdiscount.com

options-ping 1080

early-offer forced

midcall-signaling passthru

privacy-policy passthru

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

!

voice-card 0

!

crypto pki token default removal timeout 0

!

crypto pki trustpoint TP-self-signed-2714521566

enrollment selfsigned

subject-name cn=IOS-Self-Signed-Certificate-2714521566

revocation-check none

rsakeypair TP-self-signed-2714521566

!

crypto pki certificate chain TP-self-signed-2714521566

certificate self-signed 01

!

license udi pid C1861-SRST-C-F/K9 sn FCZ122760P7

file privilege 0

username admin privilege 15 password xxxx

username xxxx privilege 15 password xxxx

!

ip tcp synwait-time 10

ip ssh time-out 60

ip ssh authentication-retries 4

!

interface FastEthernet0/0

no ip address

duplex full

speed 100

!

interface FastEthernet0/0.10

!

interface Integrated-Service-Engine0/0

no ip address

!

interface FastEthernet0/1/0

description ##Vu Duo+##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/1

description ##NetGear Powerline Adaptor##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/2

description ##Sony DVD Player##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/3

description ##Qnap 210##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/4

description ##Cisco 1142##

no ip address

duplex half

speed 100

!

interface FastEthernet0/1/5

description ##Spare##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/6

description ##Spare##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/7

description ##Spare##

no ip address

duplex full

speed 100

!

interface FastEthernet0/1/8

description ##Spare##

no ip address

duplex full

speed 100

!

interface ATM0/2/0

no ip address

ip mask-reply

ip directed-broadcast

no atm ilmi-keepalive

!

interface ATM0/2/0.1 point-to-point

description ##WAN##

ip mask-reply

ip directed-broadcast

pvc 0/38

encapsulation aal5mux ppp dialer

dialer pool-member 1

!

interface Vlan1

description ##LAN##

ip address 10.0.0.253 255.255.255.0

ip directed-broadcast

ip nat inside

ip virtual-reassembly in

ip tcp adjust-mss 1452

!

interface Dialer0

description ##ADSL TALK TALK ISP##

ip address negotiated

ip mask-reply

ip directed-broadcast

ip mtu 1492

ip nat outside

ip virtual-reassembly in

encapsulation ppp

ip tcp adjust-mss 1452

dialer pool 1

dialer-group 1

ppp authentication chap callin

ppp chap hostname xxxxxx

ppp chap password xxxxxx

no cdp enable

!

ip forward-protocol nd

ip http server

ip http authentication local

ip http secure-server

ip http path flash:/gui

!

ip nat inside source list 110 interface Dialer0 overload

ip nat inside source static tcp 10.0.0.43 80 interface Dialer0 80

ip nat inside source static tcp 10.0.0.43 8080 interface Dialer0 8080

ip route 0.0.0.0 0.0.0.0 Dialer0

!

access-list 110 deny ip 10.0.0.0 0.0.0.255 14.1.1.0 0.0.0.255

access-list 110 permit ip 10.0.0.0 0.0.0.255 any

dialer-list 1 protocol ip permit

!

tftp-server flash:ip_phones

tftp-server flash:Desktops/320x212x12/CampusNight.png

tftp-server flash:Desktops/320x212x12/CiscoFountain.png

tftp-server flash:Desktops/320x212x12/MorroRock.png

tftp-server flash:Desktops/320x212x12/NantucketFlowers.png

tftp-server flash:Desktops/320x212x12/TN-CampusNight.png

tftp-server flash:Desktops/320x212x12/TN-CiscoFountain.png

tftp-server flash:Desktops/320x212x12/TN-Fountain.png

tftp-server flash:Desktops/320x212x12/TN-MorroRock.png

tftp-server flash:Desktops/320x212x12/TN-NantucketFlowers.png

tftp-server flash:Desktops/320x212x12/Fountain.png

tftp-server flash:Desktops/320x212x12/CiscoLogo.png

tftp-server flash:Desktops/320x212x12/TN-CiscoLogo.png

tftp-server flash:Desktops/320x212x12/List.xml

tftp-server flash:Desktops/320x216x16/List.xml

tftp-server flash:Desktops/320x212x16/List.xml

tftp-server flash:cvm70sccp.9-2-1TH1-13.sb alias cvm70sccp.9-2-1TH1-13.sb

tftp-server flash:term70.default.loads alias term70.default.loads

tftp-server flash:term71.default.loads alias term71.default.loads

!

control-plane

!

voice-port 0/0/0

!

voice-port 0/0/1

!

voice-port 0/0/2

!

voice-port 0/0/3

!

voice-port 0/1/0

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

voice-port 0/4/0

auto-cut-through

signal immediate

input gain auto-control

description Music On Hold Port

!

mgcp profile default

!

dial-peer voice 1 voip

destination-pattern 0T

session protocol sipv2

session target sip-server

voice-class codec 1

dtmf-relay sip-notify

!

gateway

timer receive-rtp 1200

!

sip-ua

credentials username rka61xxx password xxx realm sip.voipdiscount.com

keepalive target dns:sip.voipdiscount.com

authentication username rka61xxx password xxxx realm sip.voipdiscount.com

no remote-party-id

timers connect 100

registrar dns:sip.voipdiscount.com expires 3600

sip-server dns:sip.voipdiscount.com

connection-reuse

!

telephony-service

max-ephones 10

max-dn 10

ip source-address 10.0.0.253 port 2000

system message A Communication System

cnf-file location flash:

load 7970 SCCP70.9-2-1S.loads

max-conferences 4 gain -6

transfer-system full-consult

create cnf-files version-stamp 7960 Nov 02 2012 23:43:24

!

ephone-dn 1 dual-line

number 5001

!

ephone-dn 2 dual-line

number 5002

!

ephone-dn 3 dual-line

number 5003

!

ephone 1

description ##7945##

mac-address 0023.0434.9EDA

button 1:1

!

ephone 2

description ##7970##

mac-address 001B.5494.9CDE

button 1:3

!

line con 0

exec-timeout 20 0

logging synchronous

line aux 0

line 2

no activation-character

no exec

transport preferred none

transport input all

transport output pad telnet rlogin lapb-ta mop udptn v120 ssh

line vty 0

privilege level 15

logging synchronous

login local

transport input ssh

line vty 1 4

privilege level 15

login local

transport input none

!

ntp server 212.13.195.4

ntp server 82.219.4.30

ntp server 62.84.188.34

!

end

I would really appreciate any tips and pointers from anyone. My end goal is to have the SIP trunk registration stay up so I can make outbound calls and not have to ping to bring the trunk up after every call.

Many thanks in advance for you help

rka61 · ‎01-06-2013

Hi All

After reading some of the forums I found this discussion sounds very similar to my issue do not know if it is bug in the IOS

https://supportforums.cisco.com/thread/2170046

If someone could please confirm my suspicion.

Regards

Rushad