CALIX ONT to Adtran 908

[valunet5]

Anyone out there running the adtran 900's off of Ethernet to CALIX ONT's , to E7-20 to a Metaswitch?

Our PRI is not getting past the INVITE.

[Anonymous]

Hi,

It should work.

Please send the output of:

debug sip stack messages

debug isdn l2-formatted

Thank you,

[valunet5]

123

[Anonymous]

It looks like the Adtran is trying to register and receiving a "SIP/2.0 501 Not Implemented".

Does the Meta require registration?

Also I don't see an INVITE the logs above.

[valunet5]

Yes it does.

We have an analog line that works fine.

Thanks

[valunet5]

123

[Anonymous]

If the FXS port is working fine then can you post both the PRI and FXS config.

[valunet5]

123

[Anonymous]

Also please post the debug logs during a call from the PRI and FXS.

These may be long, if you are using putty under "Session" you will see "Logging". Here there will be an option to save all output to a file.

You should remove all private information such as passwords and IP addresses from the post above.

[david_sanchez]

Yes, you should remove all private info.  As far as your config:

1. I have a Metaswitch and I set it up both ways, with or without registration.  It depends on the customer.   I suspect you don't need to register because I see an option message received and you are getting a 501 response to the registration.  That said try removing the the following entries from your sip trunk:

   registrar primary ms.valu-net.net

   outbound-proxy primary 199.188.56.130

Make this change and just send an invite. If it fails, I would try to get the carrier to assist

2. I also noticed that your PRI interface is on T1 0/2.   I'm not sure if your TA-908e differs from ours, but on our box PRI/CAS trunks can only be on ports 3 & 4

[valunet5]

David.

Do you know what settings and values set up the talk path?

We have 4 digits passing thru the PRI to our test PBX but no talk path?

[david_sanchez]

I believe its the " media-gateway ip primary" setting under the eth interface.  Your settings look correct.  You could try using the "debug sip stack message" command in the TA-098e or a packet capture of the WAN interface to verify the SDP packets.  At that point you would verify the source and destination addresses for RTP.   If you haven't already checked the PRI side, you can use the "debug isdn l2-formatted" command to verify the B-channel and/or monitor using a PRI test set.

[david_sanchez]

FYI - I set my SIP trunk to Meta with "accept $ cost 200"

[Anonymous]

Hello and thank you for posting to our forum.

I want to make sure where you are at in your troubleshooting since there are quite a few posts on this thread.  Please correct me if I am wrong:

You can make a SIP to PRI call and it connects but you get no audio.  If no audio, is it in one or both directions?

If this is correct, here are a couple common problems.  We can look at the possibilities and come up with a plan after:

1.  There is a device in front of the ADTRAN doing NAT and it is not correctly handling the SIP and RTP ports.

2.  The RTP is not being sent to the correct destination on one or both sides.

In the case for #1, check to see if the router NATing is SIP aware and is able to properly NAT SIP messages, not just the layer 3 source and destination.

In the case for #2, we would like to see a debug of a call, which you would capture the output and either attach it to your response (I would suggest removing any sensitive info) or create a support ticket with us via phone or email.  I would run these debugs:

debug sip stack message

debug voice verbose

debug isdn l2-for

Run all 3 of these debugs from the same session so they output together.  Once the call is up, use the command: "show media sum."  This will show the active calls in the unit.  Find your call and then issue the command: "show media session 0/X.Y"  where X is the DSP and Y is the channel shown for your call in the "show media sum."  After that, stop the debug with the command "undebug all."

Regards,

Geoff

[valunet5]

GEO.

Yes we are still having problems setting up the call leg. It does the “180 ringing” accepts the call leg, and the then goes to a “clearing”

Instead of “ACK” and call setup.

Our call goes from a metaswitch to calix gpon network, that delivers 4 ethernet ports and 2 voice ports on the ONT.

The CALIX is just a fiber pipe from our customer to the metaswitch in our CO, I don’t think it is NAT’ing.

As for the second question you had. Im still trying to understand more about the “destination part” and how to troubleshoot it..

Thanks for your time on this

[Anonymous]

If the ADTRAN is going to clearing it sounds like the call is not completing.  Are you able to get a full debug:

debug sip stack message

debug voice verbose

debug isdn l2-for

Let me know if you want to continue this here or make a formal ticket in Tech Support.

Regards,

Geoff

[Anonymous]

Hello,

I went ahead and flagged this post as "Assumed Answered". If any of the responses on this thread assisted you, please mark them as Correct or Helpful as the case may be with the applicable buttons. This will make them visible and help other members of the community find solutions more easily. If you still need assistance, we would be more than happy to continue working with you on this - just let us know in a reply.

Thanks,

Geoff

[Anonymous]

Make sure your calix ont default voice vlan is not the same vlan as that of the 908. If the vlan is the same the ont will not pass the traffic to the 908. You need to create a separate profile name with the voice vlan and pass it trough to the 908. The other thing you might want to look at is the primary and secondary media gateway. The 908 pri media gateway has to match the switch's pri. media gateway others wise it will not work. I have both a Occam and a calix ont working the 924's voice traffic.