908e behind a Microtik

[curtisattravis]

I am stumped.
I am trying to set up a 908e for a SIP connection.
We are currently using a Microtik router as our gateway to the internet.

Our service provider has given us a set of public IP addresses that we can use.

Ideally, I would have liked to configure one of the 908e interfaces to use a specific public IP address, but when I tried that, I could not make it work in conjunction with the Microtik.
So, instead, I am attempting to NAT the public IP address through the Microtik router to the Adtran on an internal IP address.
I can reach the Adtran from the outside (and so could some hackers, until I tightened the services allowed in on that address.)
However, here is my problem.

When the SIP provider attempts to contact the Adtran, the Adtran is seen as "not responding back".  And I know the reason why.  When the Adtran responds back, the response is coming from the main public IP address assigned to our router, and NOT the public IP address that I have NAT'ed to the router.

I need the Adtran to respond back with the correct public IP address.

In the past, we have been using the Mitel 5000 phone system for SIP, and, handily, the Mitel has an entry line where you can enter the public NAT address that we want to be seen from that device

I can't find anything like that in the Adtran.  I see the entry for "Secondary IP address" on the interface for the port, but I am pretty sure that is not want I am looking for.

Can anyone give me any tips on this?

[jayh]

What is the WAN handoff to your ISP? If it's Ethernet with a /29 or greater subnet, insert a small switch connecting your ISP handoff, the Microtik, and the TA908e. Assign separate public IPs within the subnet to the Microtik and the TA908e.

Voice services through the TA908e don't play well if the TA908e is behind a NAT, and the TA908e has some fairly impressive IP routing capability. Can you use the TA908e as the router and ditch the Microtik, or put the Microtik behind the TA908e?

[curtisattravis]

We actually figured out how to do this with the Microtik and its NAT programming. We were not able to connect it directly to an outside address that we wanted to use, because we are using CIDR addresses.

We can actually make outbound calls now from our phone system over the T1 connection and then on through this app. I am working with the carrier about getting incoming calls to work, The calls are hitting the Adtran, but the carrier is not seeing our responses.

Get Outlook for iOS<https://aka.ms/o0ukef>

[curtisattravis]

Well, I was able to get the Microtik to mask outbound requests with the proper Public IP address, so, Hooray, I am able to register to the provider.

However....

Incoming calls just will not work.  The requests from the carrier are coming into the Adtran:(See output, more comments afterward)

(This is me, calling in from my cell phone)

16:23:53.604 SIP.STACK MSG     Rx: UDP src=208.93.135.178:5060 dst=192.168.10.249:5060

16:23:53.604 SIP.STACK MSG         INVITE sip:4053387506@174.71.189.212:5060;transport=udp SIP/2.0

16:23:53.605 SIP.STACK MSG         Via: SIP/2.0/UDP 208.93.135.178:5060;rport;branch=z9hG4bK+51038617da22d5f4dd7f5d2476ce703a1+sip+1+bbf780dd

16:23:53.605 SIP.STACK MSG        From: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>;tag=208.93.135.178+1+19f74dd4+52118c4a

16:23:53.605 SIP.STACK MSG         To: <sip:4053387506@174.71.189.212>  (This all looks good)

16:23:53.605 SIP.STACK MSG         CSeq: 23569651 INVITE

16:23:53.605 SIP.STACK MSG         Expires: 180

16:23:53.605 SIP.STACK MSG         Content-Length: 226

16:23:53.605 SIP.STACK MSG         Call-Info: <sip:208.93.135.178:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

16:23:53.606 SIP.STACK MSG         Supported: resource-priority,siprec, 100rel

16:23:53.606 SIP.STACK MSG         Contact: <sip:9e51af60bda7f139deebffe422b47e2e@208.93.135.178:5060>

16:23:53.606 SIP.STACK MSG         Content-Type: application/sdp

16:23:53.606 SIP.STACK MSG         Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name

16:23:53.606 SIP.STACK MSG         Call-ID: 0gQAAC8WAAACBAAALxYAAFleRuNMK4hl1uSUvcX4nFyP7xlRE67VaojFVisE+kvm@208.93.135.178

16:23:53.606 SIP.STACK MSG         Organization: MetaSwitch

16:23:53.606 SIP.STACK MSG         Max-Forwards: 67

16:23:53.607 SIP.STACK MSG         P-Asserted-Identity: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>

16:23:53.607 SIP.STACK MSG         Accept: application/sdp, application/dtmf-relay

16:23:53.607 SIP.STACK MSG

16:23:53.607 SIP.STACK MSG         v=0

16:23:53.607 SIP.STACK MSG         o=- 61557257777484 61557257777484 IN IP4 208.93.135.178

16:23:53.607 SIP.STACK MSG         s=-

16:23:53.607 SIP.STACK MSG         c=IN IP4 208.93.135.178

16:23:53.608 SIP.STACK MSG         t=0 0

16:23:53.608 SIP.STACK MSG         m=audio 31202 RTP/AVP 0 18 101

16:23:53.608 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

16:23:53.608 SIP.STACK MSG         a=fmtp:18 annexb=no

16:23:53.608 SIP.STACK MSG         a=silenceSupp:off - - - -

16:23:53.608 SIP.STACK MSG         a=ptime:20

16:23:53.609 SIP.STACK MSG

16:23:54.104 SIP.STACK MSG     Rx: UDP src=208.93.135.178:5060 dst=192.168.10.249:5060

16:23:54.104 SIP.STACK MSG         INVITE sip:4053387506@174.71.189.212:5060;transport=udp SIP/2.0

16:23:54.104 SIP.STACK MSG         Via: SIP/2.0/UDP 208.93.135.178:5060;rport;branch=z9hG4bK+51038617da22d5f4dd7f5d2476ce703a1+sip+1+bbf780dd

16:23:54.105 SIP.STACK MSG         From: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>;tag=208.93.135.178+1+19f74dd4+52118c4a

16:23:54.105 SIP.STACK MSG         To: <sip:4053387506@174.71.189.212>

16:23:54.105 SIP.STACK MSG         CSeq: 23569651 INVITE

16:23:54.105 SIP.STACK MSG         Expires: 180

16:23:54.105 SIP.STACK MSG         Content-Length: 226

16:23:54.105 SIP.STACK MSG         Call-Info: <sip:208.93.135.178:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

16:23:54.106 SIP.STACK MSG         Supported: resource-priority,siprec, 100rel

16:23:54.106 SIP.STACK MSG         Contact: <sip:9e51af60bda7f139deebffe422b47e2e@208.93.135.178:5060>

16:23:54.106 SIP.STACK MSG         Content-Type: application/sdp

16:23:54.106 SIP.STACK MSG         Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name

16:23:54.106 SIP.STACK MSG         Call-ID: 0gQAAC8WAAACBAAALxYAAFleRuNMK4hl1uSUvcX4nFyP7xlRE67VaojFVisE+kvm@208.93.135.178

16:23:54.106 SIP.STACK MSG         Organization: MetaSwitch

16:23:54.106 SIP.STACK MSG         Max-Forwards: 67

16:23:54.107 SIP.STACK MSG         P-Asserted-Identity: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>

16:23:54.107 SIP.STACK MSG         Accept: application/sdp, application/dtmf-relay

16:23:54.107 SIP.STACK MSG

16:23:54.107 SIP.STACK MSG         v=0

16:23:54.107 SIP.STACK MSG         o=- 61557257777484 61557257777484 IN IP4 208.93.135.178

16:23:54.107 SIP.STACK MSG         s=-

16:23:54.108 SIP.STACK MSG         c=IN IP4 208.93.135.178

16:23:54.108 SIP.STACK MSG         t=0 0

16:23:54.108 SIP.STACK MSG         m=audio 31202 RTP/AVP 0 18 101

16:23:54.108 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

16:23:54.108 SIP.STACK MSG         a=fmtp:18 annexb=no

16:23:54.108 SIP.STACK MSG         a=silenceSupp:off - - - -

16:23:54.108 SIP.STACK MSG         a=ptime:20

16:23:54.109 SIP.STACK MSG

16:23:55.104 SIP.STACK MSG     Rx: UDP src=208.93.135.178:5060 dst=192.168.10.249:5060

16:23:55.104 SIP.STACK MSG         INVITE sip:4053387506@174.71.189.212:5060;transport=udp SIP/2.0

16:23:55.105 SIP.STACK MSG         Via: SIP/2.0/UDP 208.93.135.178:5060;rport;branch=z9hG4bK+51038617da22d5f4dd7f5d2476ce703a1+sip+1+bbf780dd

16:23:55.105 SIP.STACK MSG         From: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>;tag=208.93.135.178+1+19f74dd4+52118c4a

16:23:55.105 SIP.STACK MSG         To: <sip:4053387506@174.71.189.212>

16:23:55.105 SIP.STACK MSG         CSeq: 23569651 INVITE

16:23:55.105 SIP.STACK MSG         Expires: 180

16:23:55.105 SIP.STACK MSG         Content-Length: 226

16:23:55.105 SIP.STACK MSG         Call-Info: <sip:208.93.135.178:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"

16:23:55.106 SIP.STACK MSG         Supported: resource-priority,siprec, 100rel

16:23:55.106 SIP.STACK MSG         Contact: <sip:9e51af60bda7f139deebffe422b47e2e@208.93.135.178:5060>

16:23:55.106 SIP.STACK MSG         Content-Type: application/sdp

16:23:55.106 SIP.STACK MSG         Allow-Events: message-summary, refer, dialog, line-seize, presence, call-info, as-feature-event, calling-name

16:23:55.106 SIP.STACK MSG         Call-ID: 0gQAAC8WAAACBAAALxYAAFleRuNMK4hl1uSUvcX4nFyP7xlRE67VaojFVisE+kvm@208.93.135.178

16:23:55.106 SIP.STACK MSG         Organization: MetaSwitch

16:23:55.107 SIP.STACK MSG         Max-Forwards: 67

16:23:55.107 SIP.STACK MSG         P-Asserted-Identity: "TRAVIS STRATA  " <sip:4053065047@192.168.200.40:5060>

16:23:55.107 SIP.STACK MSG         Accept: application/sdp, application/dtmf-relay

16:23:55.107 SIP.STACK MSG

16:23:55.107 SIP.STACK MSG         v=0

16:23:55.107 SIP.STACK MSG         o=- 61557257777484 61557257777484 IN IP4 208.93.135.178

16:23:55.107 SIP.STACK MSG         s=-

16:23:55.107 SIP.STACK MSG         c=IN IP4 208.93.135.178

16:23:55.108 SIP.STACK MSG         t=0 0

16:23:55.108 SIP.STACK MSG         m=audio 31202 RTP/AVP 0 18 101

16:23:55.108 SIP.STACK MSG         a=rtpmap:101 telephone-event/8000

16:23:55.108 SIP.STACK MSG         a=fmtp:18 annexb=no

16:23:55.108 SIP.STACK MSG         a=silenceSupp:off - - - -

16:23:55.108 SIP.STACK MSG         a=ptime:20

16:23:55.108 SIP.STACK MSG

16:23:55.865 SIP.STACK MSG     Tx: UDP src=192.168.10.249:5060 dst=208.93.135.178:5060

16:23:55.865 SIP.STACK MSG         REGISTER sip:208.93.135.178:5060 SIP/2.0

16:23:55.865 SIP.STACK MSG         From: <sip:4053387505@208.93.135.178:5060;transport=UDP>;tag=62852008-7f000001-13c4-1421e-b25a7e72-1421e

16:23:55.865 SIP.STACK MSG         To: <sip:4053387505@208.93.135.178:5060;transport=UDP>

16:23:55.865 SIP.STACK MSG         Call-ID: 628d9e90-7f000001-13c4-1258f-b1b5d721-1258f

16:23:55.865 SIP.STACK MSG         CSeq: 1121 REGISTER

16:23:55.866 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.10.249:5060;branch=z9hG4bK-1421e-4ea4907-7e43952e

16:23:55.866 SIP.STACK MSG         Max-Forwards: 70

16:23:55.866 SIP.STACK MSG         Supported: 100rel,replaces

16:23:55.866 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

16:23:55.866 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.6.E

16:23:55.866 SIP.STACK MSG         Contact: <sip:4053387505@192.168.10.249:5060;transport=UDP>

16:23:55.867 SIP.STACK MSG         Expires: 3600

16:23:55.867 SIP.STACK MSG         Content-Length: 0

16:23:55.867 SIP.STACK MSG

16:23:55.872 SIP.STACK MSG     Tx: UDP src=192.168.10.249:5060 dst=208.93.135.178:5060

16:23:55.872 SIP.STACK MSG         REGISTER sip:208.93.135.178:5060 SIP/2.0

16:23:55.872 SIP.STACK MSG         From: <sip:4053387506@208.93.135.178:5060;transport=UDP>;tag=628521d0-7f000001-13c4-1421e-dee327c7-1421e

16:23:55.872 SIP.STACK MSG         To: <sip:4053387506@208.93.135.178:5060;transport=UDP>

16:23:55.873 SIP.STACK MSG         Call-ID: 628d9e90-7f000001-13c4-1258f-ca6a4750-1258f

16:23:55.873 SIP.STACK MSG         CSeq: 1121 REGISTER

16:23:55.873 SIP.STACK MSG         Via: SIP/2.0/UDP 192.168.10.249:5060;branch=z9hG4bK-1421e-4ea490e-20a36b8e

16:23:55.873 SIP.STACK MSG         Max-Forwards: 70

16:23:55.873 SIP.STACK MSG         Supported: 100rel,replaces

16:23:55.873 SIP.STACK MSG         Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

16:23:55.873 SIP.STACK MSG         User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R10.9.6.E

16:23:55.874 SIP.STACK MSG         Contact: <sip:4053387506@192.168.10.249:5060;transport=UDP>

16:23:55.874 SIP.STACK MSG         Expires: 3600

16:23:55.874 SIP.STACK MSG         Content-Length: 0

The only TX messages I ever see are my Adtran sending out Registration requests to the carrier (and a lot of them, it seems!)
But I never, ever see a response to the call setup request for that incoming call.
Have I missed something in the configuration?

(see below.  We are supposed to have 4 SIP trunks to 4 PRI B Channels, with 4 DID numbers.  All passwords have been "XXX"ed out for privacy.)

!

!

! ADTRAN, Inc. OS version R10.9.6.E

! Boot ROM version R10.9.3.B1

! Platform: Total Access 908e (3rd Gen), part number 1243908F1

! Serial number LBADTN1423AF787

!

!

hostname "Adtran 908e SIP to T1"

enable password XXXXXXXXX

!

!

clock timezone -6-Central-Time

!

ip subnet-zero

ip classless

ip default-gateway 192.168.10.254

ip routing

ipv6 unicast-routing

!

!

domain-name "travisokc.com"

domain-list "travisokc.com"

name-server 8.8.8.8

!

!

no auto-config

!

event-history on

no logging forwarding

no logging email

!

no service password-encryption

!

username "admin" password "XXXXXXXXX"

!

banner motd #

Welcome to the Travis/Saddleback T1 to SIP Converter!

#

!

banner login #

You are being scanned back...

#

!

!

ip firewall

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

!

!

!

!

!

!

!

no dot11ap access-point-control

!

!

!

!

!

!

!

!

!

!

!

!

!

ip crypto ffe

!

!

!

!

!

!

!

interface eth 0/1

  description Connection to Internet

  no ip address

  no awcp

  shutdown

!

!

interface eth 0/2

  no ip address

  shutdown

!

!

!

interface gigabit-eth 0/1

  ip address  192.168.10.249  255.255.255.0

  media-gateway ip primary

  no shutdown

!

!

!

!

interface t1 0/1

  description PRI to PBX

  tdm-group 1 timeslots 1-4,24 speed 64

  no shutdown

!

interface t1 0/2

  shutdown

!

interface t1 0/3

  shutdown

!

interface t1 0/4

  shutdown

!

!

interface pri 1

  description PRI 1

  connect t1 0/1 tdm-group 1

  no 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

!

!

isdn-group 1

  connect pri 1

!

!

!

!

!

!

!

!

ip access-list standard wizard-ics

  remark Internet Connection Sharing

  permit any

!

!

ip access-list extended self

  remark Traffic to Total Access

  permit ip any  any     log

!

!

!

!

ip policy-class Private

  allow list self self

  nat source list wizard-ics interface eth 0/1 overload

!

ip policy-class Public

  ! Implicit discard

!

!

!

no tftp server

no tftp server overwrite

http server

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 local NXX-XXXX

!

!

!

!

voice codec-list Trunks

  codec g711ulaw

  codec g729

!

voice codec-list SIP

  codec g711ulaw

  codec g729

!

!

!

voice trunk T01 type isdn

  description "PRI to PBX"

  resource-selection linear descending

  connect isdn-group 1

  modem-passthrough

  t38

  rtp delay-mode adaptive

  codec-list Trunks

!

voice trunk T02 type sip

  description "SIP 01"

  sip-server primary 208.93.135.178

  registrar primary 208.93.135.178

  outbound-proxy primary 208.93.135.178

  domain "192.168.200.40"

  max-number-calls 4

  register range 4053387505 4053387506 auth-name "4053387505" password "XXXXXXXXX"

  register 4053387508 auth-name "4053387505" password "XXXXXXXXX"

  register 4053387510 auth-name "4053387505" password "XXXXXXXXX"

  codec-list SIP both

  authentication username "4053387505" password "XXXXXXXXX"

!

!

voice grouped-trunk "PRI TO PBX"

  description "PRI to PBX"

  trunk T01

  accept $ cost 0

!

!

voice grouped-trunk "SIP TO SADDLEBACK"

  trunk T02

  accept $ cost 0

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

!

ip rtp symmetric-filter

!

!

!

line con 0

  login

!

line telnet 0 4

  login

  password XXXXXXXXX

  shutdown

line ssh 0 4

  login local-userlist

  no shutdown

!

sntp server north-america.pool.ntp.org

!

ntp source gigabit-ethernet 0/1

!

!

!

end

[pmunster]

I went through a similar configuration a couple months ago on a 908 2nd gen., I needed an esbc license key to make it all work.  I am not sure about your 3rd gen, but from the product page it appears that eSBC functionality is optional so you might want to clarify this with Adtran.

[Anonymous · Accepted solution]

Try setting the domain parameter under

voice trunk T02 type sip

to be the public IP that the Mikrotik is translating the TA908 to.  I've had to do that on TA904s behind NAT.

voice trunk T02 type sip

  description "SIP 01"

  sip-server primary 208.93.135.178

  registrar primary 208.93.135.178

  outbound-proxy primary 208.93.135.178

  domain "174.71.189.212"

  max-number-calls 4

  register range 4053387505 4053387506 auth-name "4053387505" password "XXXXXXXXX"

  register 4053387508 auth-name "4053387505" password "XXXXXXXXX"

  register 4053387510 auth-name "4053387505" password "XXXXXXXXX"

  codec-list SIP both

  authentication username "4053387505" password "XXXXXXXXX"

[Anonymous]

on your voice trunk t02...

remove - domain "192.168.200.40"

and try adding one of these two...

  "sip-server secondary 174.71.189.212"

or

  "sip-server secondary 192.168.200.40"

[curtisattravis]

Ding Ding!  I owe you a beer!

That fixed my issue with the no response, and now call are routing is as they are supposed to!  Take a bow!

[curtisattravis]

One other thing I needed to set.
In the web interface, set Ringback Override 180 to "On"

Otherwise, callers will not hear ringback.

(Gee, I have learned so much in the past 50 hours!)

[Anonymous]

Great!  Took me a while to figure that one out when I ran into the same issue.    Problem: The router ignores incoming SIP messages that have URIs that don't match it's own SIP domain. Solution: change the domain to match...   Without SBC functionality, the TAs are not terribly NAT friendly.

From your sample - Domain part in bold:

16:23:53.604 SIP.STACK MSG         INVITE sip:4053387506@174.71.189.212:5060;transport=udp SIP/2.0

[curtisattravis]

This also actually addressed the original question from my ticket as well.
To solve my initial problem, I was masquerading all my traffic from the Adtran by a weird rule in my Microtik box.  This rule would send out my SIP request from local .249 as public .212.
After I changed the Domain address (that you recommended I do) and it worked for inbound, I wondered what would happen if I disabled that rule in the Microtik.  And, ta da!  I didn't need it after all, because now the Adtran is sending out those requests with the correct IP address right from the start.

Another thing I found I did not need (at least with our carrier) as I have been testing were these entries:

register range 4053387505 4053387506 auth-name "4053387505" password "XXXXXXXXX"

register 4053387508 auth-name "4053387505" password "XXXXXXXXX"

register 4053387510 auth-name "4053387505" password "XXXXXXXXX"

I don't know why I was told by an "expert" that I callled why he thought I needed these, but I did not.  I only needed the main account login credentials.

And speaking of "experts".
The main reason I started this project was to find an alternate way to deliver SIP service that we resell to customers with existing T1 interfaces.  So, on the word of both the carrier we sell for, and another company that partners with them, we bought this Adtran to test with.
And when I ran into difficulties, all of a sudden, these same sources got amnesia about how to configure it with the provider!
The conversation would go something like "Oh, yeah!  The Adtran 908!  Yeah, we installed a BUNCH of those!  Great product!"  "Great!  Do you have any write-ups on how to configure them?"  "Uh, noooooooo........"  "What about a database for an example?"  "Um, uh, no, don't have that either..."

So, you cannot imagine how grateful I am for the help you provided today.  Thanks so much!  Since I cleared this hurdle, I have been able to test this box on both a Mitel HX controller with PRI, AND a Mitel 3300/MCD with PRI, and they both work great!  (The T1 was the easy part of this whole thing!)

Now, my company has a new option to provide to our customers!

Thanks again!