I am trying to use the 908e gen2 as a sip to fxs gateway.
Here is my 908 config thus far:
!
!
! ADTRAN, Inc. OS version R11.3.0.E
! Boot ROM version 14.05.00.SA
! Platform: Total Access 908e (2nd Gen), part number 4242908L1
! Serial number CFG0612973
!
!
hostname "TA908e"
enable password xxxx
!
!
clock timezone -5-Eastern-Time
!
ip subnet-zero
ip classless
ip routing
ipv6 unicast-routing
!
!
name-server xxxx
!
!
no auto-config
!
event-history on
no logging forwarding
no logging email
!
no service password-encryption
!
username "admin" password "xxxx"
!
!
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
!
!
!
!
!
!
no dot11ap access-point-control
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
qos map ConfigWizardQoSMap 20
match dscp 46
!
!
!
!
interface eth 0/1
ip address xxxx xxxx
media-gateway ip primary
no shutdown
!
!
interface eth 0/2
no ip address
shutdown
!
!
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
shutdown
!
interface t1 0/3
shutdown
!
interface t1 0/4
shutdown
!
!
interface fxs 0/1
no shutdown
!
interface fxs 0/2
no shutdown
!
interface fxs 0/3
no shutdown
!
interface fxs 0/4
no shutdown
!
interface fxs 0/5
no shutdown
!
interface fxs 0/6
no shutdown
!
interface fxs 0/7
no shutdown
!
interface fxs 0/8
no shutdown
!
!
interface fxo 0/0
no shutdown
!
isdn-number-template 0 prefix "" plan 0 type 2 352224352X
!
!
!
!
!
!
!
!
!
!
ip route 0.0.0.0 0.0.0.0 xxxx
!
no tftp server
no tftp server overwrite
http server
no http secure-server
no snmp agent
no ip ftp server
no ip scp server
no ip sntp server
!
!
!
!
!
!
!
!
sip
sip udp 5060
no sip tcp
!
!
!
voice feature-mode network
voice forward-mode network
!
!
!
!
!
!
!
!
voice dial-plan 1 extensions 352224352X
voice dial-plan 3 local NXX-NXX-XXXX
!
!
!
!
voice codec-list Uncompressed
default
codec g711ulaw
!
!
!
voice trunk T01 type sip
description "xxxx"
sip-server primary xxxx
register tech auth-name "xxxx" password "xxxx"
!
!
voice user 3522243520
connect fxs 0/1
first-name "First"
last-name "Last"
password "1234"
no nls
rtp dtmf-relay inband
!
!
!
!
!
!
!
!
!
!
!
!
no sip registrar authenticate
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
line con 0
no login
!
line telnet 0 4
login
no shutdown
line ssh 0 4
login local-userlist
no shutdown
!
sntp server 50.7.0.147
!
!
!
!
end
The 908 has registered with the upstream Sip server (Asterisk).
The issue I am having is that whenever I dial the extension (352 224 3520) that should go to the FXS 0/1 port, I get in my debug (debug sip stack messages):
13:56:30.851 SIP.STACK MSG Rx: UDP src=xxxx:5060 dst=xxxx:5060
13:56:30.851 SIP.STACK MSG INVITE sip:3522243520@xxxx SIP/2.0
13:56:30.851 SIP.STACK MSG Via: SIP/2.0/UDP 10.8.0.1:5060;branch=z9hG4bK5dd4cae8;rport
13:56:30.852 SIP.STACK MSG Max-Forwards: 70
13:56:30.852 SIP.STACK MSG From: "WIRELESS CALLER" <sip:xxxx@xxxx>;tag=as2dc7e6f0
13:56:30.852 SIP.STACK MSG To: <sip:3522243520@xxxx>
13:56:30.852 SIP.STACK MSG Contact: <sip:xxxx@xxxx:5060>
13:56:30.852 SIP.STACK MSG Call-ID: 221b8f27796e39c04ed6282f2d892da5@10.8.0.1:5060
13:56:30.852 SIP.STACK MSG CSeq: 102 INVITE
13:56:30.853 SIP.STACK MSG User-Agent: Asterisk PBX 1.8.23.1
13:56:30.853 SIP.STACK MSG Date: Tue, 19 Aug 2014 17:56:31 GMT
13:56:30.853 SIP.STACK MSG Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
13:56:30.853 SIP.STACK MSG Supported: replaces, timer
13:56:30.853 SIP.STACK MSG Content-Type: application/sdp
13:56:30.853 SIP.STACK MSG Content-Length: 256
13:56:30.854 SIP.STACK MSG
13:56:30.854 SIP.STACK MSG v=0
13:56:30.854 SIP.STACK MSG o=root 1863201508 1863201508 IN IP4 10.8.0.1
13:56:30.854 SIP.STACK MSG s=Asterisk PBX 1.8.23.1
13:56:30.854 SIP.STACK MSG c=IN IP4 10.8.0.1
13:56:30.854 SIP.STACK MSG t=0 0
13:56:30.855 SIP.STACK MSG m=audio 14628 RTP/AVP 0 101
13:56:30.855 SIP.STACK MSG a=rtpmap:0 PCMU/8000
13:56:30.855 SIP.STACK MSG a=rtpmap:101 telephone-event/8000
13:56:30.855 SIP.STACK MSG a=fmtp:101 0-16
13:56:30.855 SIP.STACK MSG a=silenceSupp:off - - - -
13:56:30.855 SIP.STACK MSG a=ptime:20
13:56:30.856 SIP.STACK MSG a=sendrecv
13:56:30.856 SIP.STACK MSG
13:56:30.860 SIP.STACK MSG Tx: UDP src=xxxx:5060 dst=xxxx:5060
13:56:30.860 SIP.STACK MSG SIP/2.0 404 Not Found
13:56:30.860 SIP.STACK MSG From: "WIRELESS CALLER"<sip:xxxx@xxxx>;tag=as2dc7e6f0
13:56:30.860 SIP.STACK MSG To: <sip:3522243520@xxxx>;tag=4ac35f8-7f000001-13c4-62ffd-8329bdb6-62ffd
13:56:30.861 SIP.STACK MSG Call-ID: 221b8f27796e39c04ed6282f2d892da5@10.8.0.1:5060
13:56:30.861 SIP.STACK MSG CSeq: 102 INVITE
13:56:30.861 SIP.STACK MSG Via: SIP/2.0/UDP 10.8.0.1:5060;rport=5060;branch=z9hG4bK5dd4cae8
13:56:30.861 SIP.STACK MSG Supported: 100rel,replaces
13:56:30.861 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
13:56:30.862 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_2nd_Gen/R11.3.0.E
13:56:30.862 SIP.STACK MSG Content-Length: 0
13:56:30.862 SIP.STACK MSG
13:56:30.932 SIP.STACK MSG Rx: UDP src=xxxx:5060 dst=xxxx:5060
13:56:30.932 SIP.STACK MSG ACK sip:3522243520@xxxx SIP/2.0
13:56:30.932 SIP.STACK MSG Via: SIP/2.0/UDP 10.8.0.1:5060;branch=z9hG4bK5dd4cae8;rport
13:56:30.933 SIP.STACK MSG Max-Forwards: 70
13:56:30.933 SIP.STACK MSG From: "WIRELESS CALLER" <sip:xxxx@xxxx>;tag=as2dc7e6f0
13:56:30.933 SIP.STACK MSG To: <sip:3522243520@xxxx>;tag=4ac35f8-7f000001-13c4-62ffd-8329bdb6-62ffd
13:56:30.933 SIP.STACK MSG Contact: <sip:xxxx@10.8.0.1:5060>
13:56:30.933 SIP.STACK MSG Call-ID: 221b8f27796e39c04ed6282f2d892da5@10.8.0.1:5060
13:56:30.933 SIP.STACK MSG CSeq: 102 ACK
13:56:30.934 SIP.STACK MSG User-Agent: Asterisk PBX 1.8.23.1
13:56:30.934 SIP.STACK MSG Content-Length: 0
13:56:30.934 SIP.STACK MSG
I see above that it says "Not Found". The call rings busy. I'm thinking I must be missing part of the dialplan somewhere, but haven't found where to look / how to resolve it.
Any pointers?
