Hi Folks,
I've got to the point where I'm thinking of giving up on the hardware I've got because I don't think it will work, which is strange as I would have thought for a small business this would be a common configuration, unfortunately it means I'll be moving to something like DrayTek (I'd prefer to stay with Cisco though). So this is my final attempt at getting this working!
SIP TRUNK-> Cisco 877 (NAT'ing) <--> Cisco 2611XM & CME 4
- Internal calls are working
- Outgoing calls via SIP trunk and PTSN are working
- Incoming calls fail
My configuration on the CME is:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
header-passing
registrar server
outbound-proxy dns:voip.zen.co.uk
voice translation-rule 1
rule 1 /^9\(.*\)/ /\1/
voice translation-rule 2
rule 1 /.*/ /01xxxxx8882/
voice translation-profile SIP
translate calling 2
translate called 1
voice-port 1/0/1
no comfort-noise
voice-port 1/1/0
trunk-group VIC2-2FXO
supervisory disconnect dualtone mid-call
no battery-reversal
no comfort-noise
cptone GB
timeouts call-disconnect 5
timeouts ringing 90
timeouts wait-release 5
timing digit 80
timing inter-digit 80
connection plar opx 1001
!
voice-port 1/1/1
no comfort-noise
!
dial-peer voice 100 pots
trunkgroup VIC2-2FXO
description PSTN Dial Peer
destination-pattern 8T
incoming called-number .
no sip-register
!
dial-peer voice 200 voip
description SIP Outgoing Dial Peer
translation-profile outgoing SIP
destination-pattern 9T
voice-class sip outbound-proxy dns:voip.zen.co.uk
session protocol sipv2
session target dns:voip.zen.co.uk
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 150 voip
description SIP Incoming Dial Peer
voice-class sip outbound-proxy dns:voip.zen.co.uk
session protocol sipv2
session target dns:voip.zen.co.uk
session transport udp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username 01xxxxx8882 password 7 xxxxxxxxxxxxxxxxxxxxxxxxxxxx
registrar dns:voip.zen.co.uk expires 3600
sip-server dns:voip.zen.co.uk
host-registrar
permit hostname dns:voip.zen.co.uk
permit hostname dns:asterisk01.voip.zen.co.uk
permit hostname dns:asterisk02.voip.zen.co.uk
ephone-dn 1 dual-line
number 8882 secondary 012xxxx8882 no-reg both
label Andy McCall
name 012xxxx8888
call-forward noan 90xxxxxx188 timeout 30
I've taken out all of the settings from my 877, so at the moment its just NAT'ing for outgoing.
no ip nat service sip udp port 5060
ip nat inside source list 1 interface Dialer1 overload
ip route 0.0.0.0 0.0.0.0 Dialer1
ip route 192.168.2.0 255.255.255.0 192.168.1.254
!
access-list 1 permit 192.168.0.0 0.0.255.255
The things I've tried that people have suggested and I've read about:
- Adding a static route for 5060 to 192.168.1.254 to the 877 - this gets traffic to the CME, but the call doesn't ring and I get an INVALID HOST problem because the SIP trunk's IP addresses are not registered on the CME
- Adding the external IP's of the SIP provider to a loopback device - this stopped all calls working as all traffic that was ment for the SIP trunk, went to the loopback device
- Binding control and media source interface to FastEthernet0/0.1 - no changes
Please can someone help me, or is it simply a limitation of Cisco equipment that it can't work behind NAT?
Thanks!
