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?
Hello and thanks for posting to our forum.
As you know, we were able to work out this problem through Technical Support. We used the debug command - debug sip cldu alongside debug sip stack message and debug voice verbose. What we found was that the source IP address from the Asterisk SIP Invite did not match the configured sip-server ip address on voice trunk T01. The command sip-server secondary <matching source IP address from the Asterisk Invite> was used on voice trunk T01. This allowed either IP address used by the Asterisk to be recognized by the ADTRAN.
Regards,
Geoff
Hello and thanks for posting to our forum.
As you know, we were able to work out this problem through Technical Support. We used the debug command - debug sip cldu alongside debug sip stack message and debug voice verbose. What we found was that the source IP address from the Asterisk SIP Invite did not match the configured sip-server ip address on voice trunk T01. The command sip-server secondary <matching source IP address from the Asterisk Invite> was used on voice trunk T01. This allowed either IP address used by the Asterisk to be recognized by the ADTRAN.
Regards,
Geoff