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  • 1.  AT&T IP FLEX Sip Trunking

    Posted 12-15-2022 10:47 AM
    Hello IAUG,

    We recently turned up AT&T IP FLEX sip trunking. 
    We have an issue with calls not disconnecting when the far end hangs up. 
    AT&T insists this is a customer issue.

    AT&T Circuits connect to our Avaya SBCE 8.1.3.0-31-21052. 
    Inbound calls are fine.
    Outbound calls are not disconnecting when the far end hangs up. Only when we hang up does the call disconnect. 

    Wondering if anyone else is using this service and they have had similiar issues. 

    Thank you,

    ------------------------------
    Kathleen DeSantis
    Telecommunications Manager
    Maimonides Medical Center
    Brooklyn NY
    ------------------------------


  • 2.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-15-2022 10:56 AM
    Have you had a chance to look at what AT&T is sending you via the SBCE?

    You would need to run the traceSBC tool on the SBC itself, and place the outbound call.  When the far-end hangs up, you should get a BYE message from AT&T.  If you don't see that message, then there isn't much you can do.  If you do and you ignore it, you might want to talk to the group that did the Avaya side of the implementation.  Disconnect Supervision is the default setting for digital circuits... so they would have had to do something odd to turn it off.

    The only thing to note is don't test with old POTS connections.  Disconnect supervision is sketchy at best when calling POTS/landline customers.

    ------------------------------
    Nick Kwiatkowski
    Director of Design and Engineering
    Michigan State University
    East Lansing MI
    ------------------------------

    Original Message


  • 3.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-15-2022 11:15 AM
    Thank you Nick.

    We did traceSBC and there was no BYE.
    We've been trying for more than a few days to get AT&T to work this issue. 
    Does anyone else have AT&T or a person at AT&T who will work this - we are getting a lot of pushback.
    We are currently waiting (24 hour wait time) to engage with a network engineer from AT&T.

    ------------------------------
    Kathleen DeSantis
    Telecommunications Manager
    Maimonides Medical Center
    Brooklyn NY
    ------------------------------

    Original Message


  • 4.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-15-2022 11:27 AM
    Simply telling the engineers that you aren't getting a BYE message at the end of a call should be enough.  That's basic SIP 101 and is mandatory in the specs. 

    All that being said, it's not a big surprise.  When we initially tried to implement AT&T IP Flex on our campus we had nothing but problems, primary with them not following the specs that even they put out for their own product.  Things like /sometimes/ they would send G.723a instead of G.711u like we were supposed to be locked down with, or them sending FORBIDDEN messages when we tried to transfer calls or put calls on hold.  Some of those things can be fixed with an SBC, but there were enough small things that piled up that made us abort the project.

    -Nick

    ------------------------------
    Nick Kwiatkowski
    Director of Design and Engineering
    Michigan State University
    East Lansing MI
    ------------------------------

    Original Message


  • 5.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-16-2022 01:26 AM
    If your signaling is using TCP, try setting the SBC, in the SIP Server settings for the AT&T profile, to send options every 90 seconds.  We have one carrier that has a firewall that silently closely the TCP session if it idles more than 2 minutes, preventing their own SBC from sending us session refresh reinvites.  This would cause the call the die without a BYE. 





    Original Message


  • 6.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-16-2022 08:06 AM

    We have been using AT&T IPFlex for 12 years and seen similar weird problems.  

    I do not want to make assumptions on who owns the router, that terminates the data circuit that BVoIP is running on.  I've learned that BVoIP is the service on the circuit and IPFlex or Enhanced IPFlex, is considered an application.  We started out using a router they provided at first and about a year later, we provided our own.  It was part of our learning curve.

     

    Odd that TCP is working because their documentation says they only support UDP.   Wonder if the terminating router, is doing UDP back them and TCP to the Avaya SBC, Interesting.

     

    I bugged my AT&T account person for months to get my CODEC straight. Typically, the only reason they do anything less than G.729 is the concurrent call capacity planning, dictated G.723.

     

    When I do have to submit trouble ticket to AT&T, they magically forget to tell you is, provide 3 or more call examples, EVERY DAY.  Seems their backend suffers from short term memory loss.

     

    Finally, pull a packet capture. They love those. If you can get one at or after the router and before the SBC, it will help prove your case.

     

    Hope this helps. 

    Alan

     

     

    Alan W. Williford
    Telecommunications Scientist

    Global Network Communications

    Liberty Mutual Insurance

     




    Original Message


  • 7.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-16-2022 10:21 AM
    We also have IP Flex installed, but as a PRI handoff due to the fact that we are still on CM 6.3.  For the most part, things work fine.  However, there are few stations that are experiencing dropped calls.  When a caller dials in they can hear our end-user.  Our end-user cannot hear the calling party.  After a few seconds the call drops and a message plays "your call cannot be completed...hang up and dial *611".  For some reason the call bounces out.  After doing a trace on a call that they placed, AT&T states that the PBX is kicking the call back out.  They suggested rebuilding the station.  I did that and we get the same problem.  I am not familiar with reading their traces.  I attached one of our traces and their trace.  Does anyone see anything that could be wrong on our end?

    ------------------------------
    Susan Cope
    Telecommunications Manager
    McCormick Place
    Chicago, Illinois
    312-791-6536
    scope@mccormickplace.com
    ------------------------------

    Attachment(s)

    pdf
    7203 test 12_16_22.pdf   4 KB 1 version
    pdf
    ATT CCBCAD (002).pdf   129 KB 1 version
    Original Message


  • 8.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-16-2022 10:57 AM
    Most of the times I've seen that type of scenario are when one of the two sides proposes switching to a CODEC that isn't supported on the other side.

    For example, you might support g.711u and g.722k, and your endpoints are programmed to pick g.722k first.  A call comes in as g.711u and after 2 seconds when the phone picks up, CM tells AT&T via the SBC that you are switching to g.722k.  AT&T throws you the virtual middle finger (via a 500 message) and hangs up the call.    On our system, we only support g.711u, OPUS and g.722k,  and sometimes our carrier sends calls with other codecs.  Our choice is to either drop the call, propose a new one and hope they change, or transcode on our side.

    This can usually be found in the SIP traces on the SBC.  In theory the SBC should be flagging those calls that get forcibly dropped because of codec issues, but it often ends up being too noisy to be easy to identify those issues.

    ------------------------------
    Nick Kwiatkowski
    Director of Design and Engineering
    Michigan State University
    East Lansing MI
    ------------------------------

    Original Message


  • 9.  RE: AT&T IP FLEX Sip Trunking

    Posted 12-16-2022 11:26 AM
    Checking the CODEC sets, I see only G.711MU.  90% of our phones are old digital sets (6400-series).  The stations that are experiencing this problem are 6400-series phones.  It's odd that this is only happening to a few stations, while others work just fine.  I'm lost on this one.

    ------------------------------
    Susan Cope
    Telecommunications Manager
    McCormick Place
    Chicago, Illinois
    312-791-6536
    scope@mccormickplace.com
    ------------------------------

    Original Message