Adtran is responding with 501 method not implemented for Register request.

[hnshiva]

Hi All,

I am sending Register messages from our MiVoice Business to one of our service provider through Adtran 908e.

MiVoice Business is sending Register messages as required by the service provider,

but Adtran is responding with 501 method not implemented.

Please find the below SIP messages. What could be the possible reason? Please suggest.

REGISTER sip:voip.ITSPlink.com:5100 SIP/2.0

Via: SIP/2.0/UDP xxx.xxx.xx.xx:5060;branch=z9hG4bK3064070084-4035936290

Route: <sip:xxx.xxx.xx.xx:5060;transport=udp;lr>,<sip:xxx.xxx.xx.xx:5100;lr>

Max-Forwards: 70

Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER,UPDATE

From: <sip:6023471866@xxx.xxx.xx.xx>;tag=0_3064070108-4035936291

To: <sip:6023471866@xxx.xxx.xx.xx>

Call-ID: 3064070017-4035936289

CSeq: 1 REGISTER

Expires: 3600

Contact: <sip:6023471866@xxx.xxx.xx.xx:5060;transport=udp>

User-Agent: Mitel-3300-ICP 20.0.0.171

Content-Length: 0

SIP/2.0 501 Not Implemented

From: <sip:6023471866@xxx.xxx.xx.xx>;tag=0_3064070108-4035936291

To: <sip:6023471866@xxx.xxx.xx.xx>;tag=4fb23538-7f000001-13c4-b3-e6586a86-b3

Call-ID: 3064070017-4035936289

CSeq: 1 REGISTER

Via: SIP/2.0/UDP xxx.xxx.xx.xx:5060;branch=z9hG4bK3064070084-4035936290

Content-Length: 0

!======================================================================

!

! 1 Trunk connection for each trunk group configured at ITSPLink

! Authentication name and password as supplied by ITSPLink

!

!======================================================================

!

voice trunk T01 type sip

  description "SIP to ITSP One"

  sip-server primary xxx.xxx.xx.xx udp 5100

  registrar primary xxx.xxx.xx.xx udp 5100

  domain "xxx.xxx.xx.xx"

  hmr Trunk01_Outbound out

  register 6023471866

  codec-list g711u

  authentication username "xxxxxxxxxxx" password "xxxxxxx"

!

! !======================================================================

!

! 1 Trunk connections to the MiVoice Business

!

!======================================================================

!

!

voice trunk T02 type sip

  description "SIP_to MiVoice Business"

  sip-server primary xxx.xxx.xx.xx

  codec-list g711u

!

!

voice grouped-trunk ITSP

  description "ITSP sip trunk"

  trunk T01 

  accept $ cost 0

!

!

voice grouped-trunk MITEL

  description "mitel MiVB"

  trunk T02

  accept NXX-NXX-XXXX cost 0

  accept 1-NXX-NXX-XXXX cost 0

  accept 011-$ cost 0

  accept 411 cost 0

  accept 611 cost 0

  accept 911 cost 0

  accept *$ cost 0

  accept $ cost 0

!

[Anonymous]

Hi hnshiva,

Looks to me that your Adtran is set up as a back-to-back user agent (B2BUA), as opposed to a proxy, so the behavior you're seeing is by design: the 908 registers to your ITSP on behalf of the PBX, therefore a separate registration from Mitel to 908 (or ITSP for that matter) is not needed.

Consider this diagram:

[ ITSP ]    <== Segment 1 ==>  [ 908 voice trunk T01]   ||   [908 voice trunk T02 ]  <== Segment 2 ==> [ Mitel PBX ]

Traffic received on Segment 1 is terminated on the 908. When needed (i.e. if it's an inbound call attempt) the traffic will be re-generated on Segment 2 and sent to the PBX.

The same is true in reverse. Segment 2 traffic will be sent to Segment 1 only if it's a call attempt. Registration traffic from PBX will not be sent to Segment 1, since Adtran is already handling that "conversation" with the ITSP. Any registration attempts here will be greeted with the SIP 501 error.

On the 908, verify that the SIP account is correctly registered by issuing a "show sip trunk-reg" command at the CLI. You should see a small table with your registered user (6023471866) along with Yes / No for registration status, and other registration statistics. As long as that line is registered, you can disable registrations on your Mitel. When calls (INVITEs) are received from Mitel's IP, the 908 will send those northbound to your ITSP. If challenged (SIP 401 message sent by ITSP) the Adtran will send back voice trunk 01 SIP user's credentials and calls will proceed.