HELP!!! eth0/1 to eth0/2 SIP SBC with 908e?????

[rduncan]

I have a Adtran 908e that I need to create a very simple SBC from SIP on eth0/1 to SIP on eth0/2... I am having problems and can not find anything on doing this.

thanks!!!!

[jayh]

Do you need true SBC functionality or are you just needing NAT traversal and/or transparent proxy.

Perhaps if you were to describe a bit more specifically what you are trying to accomplish it would help.

[rduncan]

I just need a trans proxy... I just want to pass the SIP traffic through from eth0/1 to eth0/2.... Sorry for being vague. Thanks for any help!!

[rduncan]

This is between our PBX and the telephone company.

[jayh]

Take a look at this for basics.  https://supportforums.adtran.com/docs/DOC-1826

Is the unit in place now and you are adding SIP, or is it a new installation?  Will the TA908e be doing NAT for the LAN?  Are there IP phones connected to the PBX now, on the LAN?  The more information you give the community, the more help we can be. 

[rduncan]

We currently have an openscape pbx going to a mediatrix the goes pri to the adtran that goes to the telecom... The mediatrix is freexing and taking out the phones daily... I am trying to find a cheap and easy way to get connectivity between us and the telecom via sip to sip connection... The current method is sip to pri to sip which is not good...

I think that the openscape is failing to connect to the adtran as I got it to ring by just pluggin it into the adtran and calling in and out... Though there was not voice either way... Once rebooting the adtran the calling is now saying error.

thanks,

[rduncan]

Is there a way to register the adtran to the PBX? The PBX does not seem to want to send anything unless the endpoint is registered.

thanks!!

[jayh]

Yes, you can register the Adtran to the PBX (or the PBX to the Adtran if that's what you need).

It sounds like the best configuration for your needs is to create two SIP trunks on the Adtran, one pointing to the carrier and one to the PBX.  Then build voice grouped-trunks to steer the calls appropriately.

Each SIP trunk can be configured separately with whatever registration, codec, etc. is required by the other end.

This is different from transparent proxy which would be for a SIP phone going through the TA908e to a hosted PBX as an example.

[rduncan]

Sorry for my lack of knowledge, I did not put the system in, nor am I a telecoms guy lol... Story of every IT persons life I guess...

Well, I have setup what you stated, two trunk groups and two trunk accounts routing to the sip servers. Do you know what the command to register the adtran to the pbx is? I don't seem to be figuring it out. I pasted the config below.

ip firewall

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

no dot11ap access-point-control

!

ip flow top-talkers

!

interface eth 0/1

  description Internal

  ip address  10.4.8.84  255.255.255.192

  media-gateway ip primary

  no awcp

  no shutdown

!

!

interface eth 0/2

  description Ridgeville

  ip address  192.168.254.3  255.255.255.0

  media-gateway ip primary

  no awcp

  no shutdown

!

!

!

!

interface t1 0/1

  shutdown

!

interface t1 0/2

  shutdown

!

interface t1 0/3

  tdm-group 1 timeslots 1-24 speed 64

  shutdown

!

interface t1 0/4

  tdm-group 1 timeslots 1-24 speed 64

  shutdown

!

!

interface pri 1

  description Mediatrix-Top-PRI-2

  connect t1 0/3 tdm-group 1

  no shutdown

!

interface pri 2

  description Mediatrix-Bottom-PRI-2

  connect t1 0/4 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

!

!

interface fxo 0/0

  no shutdown

!

!

isdn-group 1

!

ip access-list extended ridgeville

  permit ip any  any     log

!

ip access-list extended web-acl-2

  remark ping

  permit icmp any  any  echo   log

!

!

!

!

ip policy-class management

  allow list mgmt self stateless

!

ip policy-class ridgeville

  allow list web-acl-2 self

  allow list taco self stateless

  allow list ridgeville

!

!

!

ip route 0.0.0.0 0.0.0.0 10.4.8.65

ip route 192.168.2.0 255.255.255.0 192.168.254.1

!

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

sip tcp 5060

!

!

!

voice feature-mode network

voice forward-mode network

voice conferencing-mode local

!

!

!

!

!

!

!

!

voice dial-plan 1 local NXX-NXX-XXXX

!

!

!

!

voice codec-list PERMITTED_CODECS

  default

  codec g711ulaw

!

!

!

voice trunk T01 type sip

  description "Ridgeville"

  sip-server primary 192.168.2.2

  registrar primary 192.168.2.2

  codec-list PERMITTED_CODECS both

  update-supported

!

voice trunk T02 type sip

  description "OpenScape"

  sip-server primary 10.4.8.54 tcp

  registrar primary 10.4.8.54 tcp

!

!

voice grouped-trunk RIDGEVILLE

  description "SIP Trunk to Ridgeville"

  trunk T01

  accept $ cost 0

!

!

voice grouped-trunk OPENSCAPE

  description "PRI to OpenScape PBX"

  trunk T02

  accept $ cost 0

!

!

!

!

!

!

!

!

!

!

!

!

no sip registrar authenticate

!

!

sip proxy

sip proxy transparent

!

!

!

!

!

!

!

!

sip grammar from host local

sip grammar p-asserted-identity host local

!

!

!

!

no ip rtp firewall-traversal reuse-nat-ports

!

!

sip secure remote-user

  no blacklist

!

!

ip rtp quality-monitoring

ip rtp quality-monitoring udp

ip rtp quality-monitoring sip

!

[jayh]

OK, what is working so far?  Is "Ridgeville" the incoming telco?  When you call the PBX number from the outside do you see incoming SIP invites?  Things are pretty confusing as both sides have private addresses.  Your default route seems to be on the subnet that you have labeled as "Internal", which seems odd.

Is there a device at your site that is doing NAT between the incoming SIP trunk and the Adtran?

Does the PBX expect to register to something, or does it want the incoming trunk to register to it? 

You may need to get either a local consultant or someone from Adtran ACES to help you with this. 

[rduncan]

I have worked with Adtran and we have gotten it kinda working. We now have it routing calls in and out but voice traffic does not get routed out properly as there is no outgoing voice and any outgoing calls drop after 16 seconds. Very sketchy. The routing looks good, so we are stuck at that.... I am now looking at Yate and openSIP for alternatives to the SBC or anything to proxy the traffic.

[Anonymous]

Hello,

I looked for your ticket today here at ADTRAN and I found it.  Have you been able to figure out what was changing the source IP of the SIP messages?  If the issue is not resolved, just let me know and I can look into it further. 

Thanks,

Geoff

[rduncan]

We have not found out what is going on. We also have the call on outbound calls dropping after 16 seconds.

[Anonymous]

rduncan,

     Do you have the SBC version of the TA 900?  If not I do not believe what you are trying will work properly. If so try adding:

!

ip rtp symmetric-filter

ip rtp media-anchoring

!

Media-Anchoring keeps the TA unit in the middle of the call which is needed since you are routing the traffic between the carrier and the PBX.

One other thing I would recommend is not having both trunks set as accept $ cost 0.  I would only allow the actual numbers on the PBX side under voice grouped-trunk OPENSCAPE that will make it a little easier for the switchboard to route the calls to the proper trunk.

[rduncan]

We do not have the SBC version so those commands do not work... We have a 908e without SBC

[Anonymous]

rduncan,

    To the best of my knowledge and experience it requires the SBC version of the Adtran 908e/SBC in order to do what you are trying to do.  Maybe someone else has a work around but I have never been able to get it to work without using the SBC because you need media anchoring in order to keep the RTP channel open otherwise when the 908e steps out of the call the two end points cant talk anymore and the rtp stream breaks.

I would recommend using an Adtran 3430 SBC instead as long as you need to do any TDM handoff (like analog).  The 3430 SBC is lower cost then the 908e/SBC.

John Wable

[rduncan]

Thanks man, I guess I will either buy one or find something else. I am working with openSIPS to see if they can get it to work.

thanks again!!!

I will update if I ever get it to work.

[Anonymous]

Hello Robert,

I was wondering if you are still working on this and if you made any progress.  Let us know if you have any questions.

Thanks,

Geoff