TA924e Incoming Call Routing SIP to PRI and SIP to FXS

[TPS_GBD]

Hello Community,

I am configuring an Adtran 924e 2nd Gen with Cloud-Based SIP trunk connecting to Premise Based PBX with PRI, and FXS ports for Fax Machines.

1. SIP trunk is registered and operational.
2. SIP Voice Trunk Group is configured, and SIP trunk account is assigned.
3. PRI trunk is connected to the PBX and is In-service and operational.
4. ISDN Trunk Group is configured, and PRI trunk account is assigned.
5. 6 Analog Voice Users are assigned to FXS Ports 1 through 6.
6. 10-digit dial plan is assigned
7. Static IP Address, default gateway, DNS, NTP address, Default routing is configured.

We are able to place any outgoing call from the PBX out the PRI to the PSTN via the SIP trunk successfully. We are also able to place any outgoing call from any of the FXS ports successfully.

We have one test number assigned to the SIP trunk and when we dial the test number the far end gets a busy signal. We need help routing incoming calls Attached to this post is a SIP debug.

I see some logical information in the debug, (404 not found) which suggests that we do not have a desitination matching the dialed number in the SIP request, but I do not know where to go to resolve it. I have created an analog voice user matching the DID, I have created an alias for an existing analog voice user assigned to an FXS port, I have built an incoming route in the PBX to accept the DID if a call is delivered to the PRI trk from the SIP trk, but nothing works.

Thanks in advance for any assistance!

[geo]

Hello TPS_GBD,

Thanks for posting to the support community!  A couple things to check:

INVITE sip:6105551212@172.56.220.190:55406 SIP/2.0

This is the request-URI on the Rx call.  Does the information after the @ match what is configured in the SIP voice trunk?  Or can that be resolved from a configured sip-server FQDN?
Try running these debugs for more information:

debug sip stack message
debug sip cldu
debug voice verbose
debug sip syntax
debug sip stack level error

Make the call again and see if something sticks out in the debug.  From what I can tell the 924e doesn't like the formatting of the request-URI.

Thanks!

Geo

[TPS_GBD]

Adding the configuration here with customer specific information and telephone numbers changed for security reasons.

[geo]

Hello,

Going off the request-URI, which looks to be why the inbound call is failing, INVITE sip:6105551212@172.56.220.190:55406 SIP/2.0 - note the IP address 172.56.220.190.

That IP does not match - outbound-proxy primary 162.252.250.41

I am unable to resolve this - sipcust.voip.phaze-2.cloud, so you would need to enter the command: show host and check it there is a 172.56.220.190 there, but I doubt it.  It appears that there is a device on the edge in front of the TA924e that is not properly NATing SIP.  It might be doing the source and destination L3 address, but it is not NATing the actual SIP packet.  The device would need a proper SIP ALG to do so.

Based on the information that you can make outbound calls, but inbound calls are falling in this manner, I would look into the device in front of the ADTRAN that is NATing.

Thanks,

Geo