1 Reply Latest reply on Aug 20, 2014 8:59 AM by geo

    SIP -> FXS on a 908E gen2

    bbb New Member

      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?

        • Re: SIP -> FXS on a 908E gen2
          geo Employee

          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