I have a basic configuration with SIP provider via eth 0/1 and a PRI connected with a PBX.
The unit is installed behind a NAT router with the public IP 108.X.X.X.
I can't receive incoming calls and in debug I get:
12:50:10.688 SIP.STACK MSGSUM Tx: REGISTER sip:64.XX.XX.XX:6060 SIP/2.0
12:50:10.710 SIP.STACK MSGSUM Rx: SIP/2.0 200 OK
12:50:11.102 SIP.STACK MSGSUM Rx: INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:12.305 SIP.STACK MSGSUM Rx: INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:12.906 SIP.STACK MSGSUM Rx: CANCEL sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0
12:50:12.941 SIP.STACK MSGSUM Rx: INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:13.509 SIP.STACK MSGSUM Rx: CANCEL sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0
12:50:13.741 SIP.STACK MSGSUM Rx :INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:14.711 SIP.STACK MSGSUM Rx: CANCEL sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0
12:50:14.944 SIP.STACK MSGSUM Rx: INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:15.545 SIP.STACK MSGSUM Rx: CANCEL sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0
12:50:15.647 SIP.STACK MSGSUM Rx: INVITE sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0 (with SDP)
12:50:15.719 SIP.STACK MSGSUM Tx: REGISTER sip:64.XX.XX.XX:6060 SIP/2.0
12:50:15.741 SIP.STACK MSGSUM Rx: SIP/2.0 200 OK
12:50:16.147 SIP.STACK MSGSUM Rx: CANCEL sip:973NNNNNNN@108.X.X.X:1954;tgrp=idxxxxx SIP/2.0
I can make outgoing calls.
This is the configuration
ip subnet-zero
ip classless
ip default-gateway 192.168.254.1
ip routing
!
!
ip domain-proxy
!
!
!
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
no dot11ap access-point-control
!
interface eth 0/1
ip address 192.168.254.39 255.255.255.0
media-gateway ip primary
no shutdown
!
!
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
description Mitel PRI
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
!
interface pri 1
description pri 1
isdn name-delivery setup
connect t1 0/2 tdm-group 1
role network b-channel-restarts disable
no shutdown
!
isdn-group 1
connect pri 1
!
timing-source internal
!
ip route 0.0.0.0 0.0.0.0 192.168.254.1
ip route 192.168.254.0 255.255.255.0 192.168.254.1
!
!
ip sip
ip sip udp 5060
no ip sip tcp
!
!
voice feature-mode network
voice forward-mode network
!
!
voice dial-plan 1 local NXX-NXX-XXXX
voice dial-plan 2 long-distance 1NXXXXXXXXX
!
!
voice codec-list Main
codec g729
codec g711ulaw
!
!
!
voice trunk T01 type sip
description "Intermedia SIP"
sip-server primary 64.X.X.X udp 6060
registrar primary 64.X.X.X udp 6060
authentication username "idxxxxx" password "1234"
max-number-calls 10
register 1 auth-name "idxxxxx" password "1234"
codec-group Main
!
voice trunk T02 type isdn
description "PRI to Mitel"
resource-selection circular descending
connect isdn-group 1
rtp delay-mode adaptive
!
!
voice grouped-trunk SIP
trunk T01
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 1-800-NXX-XXXX cost 0
accept 1-888-NXX-XXXX cost 0
accept 1-877-NXX-XXXX cost 0
accept 1-866-NXX-XXXX cost 0
accept 1-855-NXX-XXXX cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept 1-900-NXX-XXXX cost 0
accept 1-976-NXX-XXXX cost 0
accept NXX-976-XXXX cost 0
!
!
voice grouped-trunk "PRI TO MITEL"
trunk T02
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 1-800-NXX-XXXX cost 0
accept 1-888-NXX-XXXX cost 0
accept 1-877-NXX-XXXX cost 0
accept 1-866-NXX-XXXX cost 0
accept 1-855-NXX-XXXX cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept 1-900-NXX-XXXX cost 0
accept 1-976-NXX-XXXX cost 0
accept NXX-976-XXXX cost 0
!
end
Your debugs show the INVITE coming from a 108.X.X.X address. Your unit isn't configured to accept SIP from this IP. Can you modify the settings of the NAT router to not manipulate the source address? I assume that the INVITE is originating from the 64.X.X.X Intermedia SIP server, correct?
Alternatively, configure a SIP trunk T03 with the SIP server of whatever 108.X.X.X address your NAT router is using for origin. It shouldn't need to register or need a grouped-trunk if only incoming calls appear from that address.
I would strongly advise against putting the TA900 behind a NAT device. Doing so generally results in some form of random breakage. Put it on a public IP address. Stuff like you're seeing now, one-way audio, etc. will drive you batty.
For your PRI TO MITEL grouped-trunk, it's probably best to only accept the DID range(s) assigned to the PBX instead of everything. You definitely don't want that trunk accepting 911 (unless it is at a PSAP which I doubt). You also probably don't want any trunk accepting 1-900-NXX-XXXX or 1-976-NXX-XXXX or NXX-976-XXXX.
64.X.X.X is the SIP server and 108.X.X.X is my public IP.
This is only an evaluation unit and that is the reason some stuff is left in the programming.
I know that by connecting it to the public IP will work, but I was looking to see if it is a place where like with other devices,
you can enter the public IP address, to be used for this setup.
Your suggestion to setup another SIP trunk is very clever but looks that at least in the call setup phase, is using two SIP connections.
I found the solution.
I put the public IP address in voice trunk T01 ( my SIP trunk) for the domain.
voice trunk T01 type sip
domain "108.X.X.X"