PRI to call manager

[Anonymous]

trying to interface sip from a Meta switch to a Adtran 904e and convert to PRI to talk to a call manager. Sip works fine to 904 can not get the PRI side of 904 to talk to the Call Managero Call

[Anonymous]

Farmers,

Thanks for posting!  When you are having trouble placing calls from SIP to PRI on a Total Access 900e series, you may want to first verify you have the basic configuration as shown below.

interface t1 0/4

  tdm-group 1 timeslots 1-24

  no shutdown

!

interface pri 1

  connect t1 0/4 tdm-group 1

  no shutdown

!

isdn-group 1

  connect pri 1

!

voice trunk T02 type isdn

  connect isdn-group 1

!

voice grouped-trunk PRI

  trunk T02

  accept $

!

! SIP TRUNK CONFIGURATION

!

voice trunk T01 type sip

  sip-server primary <INSERT_SIP_SERVER_ADDRESS>

!

voice grouped-trunk SIP

  trunk T01

  accept $

Once you have verified this all appears correct, I would then check the output of "show interfaces" to verify that all interfaces are up.  In particular, verify that you have an active D channel on the PRI.  An easy way to check the D channel is in Channel status portion of the output of "show interface pri 1".  You should see a 'D' under the last channel.

If the D channel is up, we will likely need to check debug output to determine if the call is received by the ADTRAN unit and how that call is routed.  Below are the typical debug commands we would enter, and then place a test call.

• debug sip stack message    // This will verify the SIP message for the inbound call reaches the unit.
• debug sip cldu                    //  This will verify that the SIP message is validated by the CLDU.
• debug voice summary         // This output will often give a simple explanation call flow.
• debug voice verbose            // This will show internal processing and often requires ADTRAN Technical Support review.
• debug isdn L2-formatted      // This shows D channel messages on the PRI.

If you need assistance with the interpretation of this debug output, you may need to create a normal Technical Support ticket by calling 888-423-8726.  You can respond on this forum, but we want to make sure that no sensitive information, such as IP addresses, phone numbers, etc., are shown to everyone.

Thanks again,

David

[Anonymous]

David the inbound calls worked fine but still could not call out here is the capture from the Adtran could look over and  see what might be the problem Thanks Rodney - ticket #1334465

Message was edited by: david.  Removed customer phone number and possibly sensitive information included in the debug capture.

[Anonymous · Accepted solution]

Rodney,

The outbound call is failing because the softswitch rejects the call with the following SIP message.

22:28:12.152 SIP.STACK MSG         SIP/2.0 403 From: URI not recognized

You may need to check the softswitch logs to be certain, but often you get this type of reject message because the caller ID isn't an expected value.  In your case, the ADTRAN unit sends out "3305" as the caller ID because that is what is being received from the PBX.  Below are a couple of things I would try.

1. Change the PBX settings so that it sends out the proper 10-digit DID associated with that site for the CALLING PARTY # (caller ID).  The ADTRAN unit will then automatically map that value into the From: header in the SIP packet.

2. Override the caller ID on the ADTRAN unit.  Below is an example.

int pri 1

     calling-party override always

     calling-party number <10-digit caller ID associated with this site>

3. You could also modify the caller ID outbound on SIP trunk as shown below.

voice trunk T01 type sip

     match ani $ substitute <10-digit caller ID associated with this site>

I would prefer the first option as it allows the PBX to use multiple 10-digit caller ID numbers that might be allowed by the softswitch.  The second and third options would change the caller ID, but only one number would be used for caller ID for every user behind the PBX.

If changing the caller ID number does not resolve the issue, check the softswitch to determine the cause of the 403 message sent out.  We may need to modify SIP grammar on the host portion in the From: header.

Thanks,

David

ADTRAN Technical Support

[Anonymous]

Rodney,

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 answers 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, I would be more than happy to continue working with you on this - just let me know in a reply.

Thanks,

David

ADTRAN Technical Support

[Anonymous]

We just published a new guide that details which configuration options are used to determine outbound caller ID:

Outbound Caller ID priority in AOS Voice products

Thanks,

Matt

[Anonymous]

Farmers,

I went ahead and flagged the "Correct Answer" on this post to make it more visible and help other members of the community find solutions more easily. If you don't feel like the answer I marked was correct, feel free to come back to this post and unmark it and select another in its place with the applicable buttons.  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,

David