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New Contributor II

SIP -> FXS on a 908E gen2

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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?

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Anonymous
Not applicable

Re: SIP -> FXS on a 908E gen2

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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

View solution in original post

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1 Reply
Anonymous
Not applicable

Re: SIP -> FXS on a 908E gen2

Jump to solution

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

View solution in original post

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