Cisco Phones call outbound, but don't receive inbound

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Brandon B

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Sep 16, 2016, 12:29:57 AM9/16/16
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Hey Group,

Hoping someone out there has seen this one.  I think we have the problem pretty well figured out, just don't know how/where to fix.  These are 7940/60 phones.  I know.  I'm sorry.  Shame. 

Phones register no problem.  Outbound calls no problem....except, if the external party hangs up first, the bye is sent and the Cisco responds with a 404.  Incoming calls do the same - call comes in, hits phone, 404 Not found.

Did some comparisons against asterisk, and the difference appears to be in the request-URI.  Coming from Kazoo it is using the external IP and port, asterisk is sending the internal IP and port.  The Cisco is "trained" to look at the request-URI and if it doesn't match it gives a 404 Not Found and doesn't answer (or doesn't hang up).  (more on that here: https://supportforums.cisco.com/discussion/10198046/buggy-nat-behaviour-7960-pos3-08-8-00-sip-fimrware)  **side note, I have tried versions 8-5, 8-6, 8-7, 8-8, 8-12 of the firmware.  

We looked at the NDLB in freeswitch and Nat settings in Kamailio, but didn't find any smoking guns.  

Our setup:
FSW: 1.4.15
Kml: 4.3.4
Kazoo: 3.22.50

Apparently pastie.org (wth!!) is down, so here is a quick log deal (I suppose asterisk also sends proxy auth which is different?)

Kazoo Bye (which gets rejected)

Session Initiation Protocol (BYE)
    Request-Line: BYE sip:867@EXTERNALIPOFPHONE:47894;user=phone;transport=udp SIP/2.0
    Message Header
        Via: SIP/2.0/UDP KAMAILIO:5060;branch=z9hG4bK1d8c.d024c8e7c11b4aa463be0af25e9015cc.0
        Via: SIP/2.0/UDP FREESWITCH:11000;received=54.148.57.6;rport=11000;branch=z9hG4bKB197XjSKQB1BS
        Max-Forwards: 50
        From: <sip:1237@REALM>;tag=9Sa02DmUgvBFS
        To: "867" <sip:867@REALM>;tag=00070e56ecee0009207708af-1d4c4d03
        Call-ID: 00070e56-ecee0004-79a4257f-2b6ba599@INTERNALIPOFPHONE
        CSeq: 96661284 BYE
        User-Agent: US
        Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
        Supported: path, replaces
        Reason: Q.850;cause=16;text="NORMAL_CLEARING"
        Content-Length: 0

Asterisk Bye (which works fine)

Session Initiation Protocol (BYE)
    Request-Line: BYE sip:867@INTERNALIPOFPHONE:5060;user=phone;transport=udp SIP/2.0
    Message Header
        Via: SIP/2.0/UDP ASTERISKIP:5060;branch=z9hG4bK2c0873b0;rport
        Max-Forwards: 70
        From: <sip:5555555555@ASTERISKREALM>;tag=as157e93c6
        To: "867" <sip:867@ASTERISKREALM>;tag=00070e56ecee003266d008f2-55220dc7
        Call-ID: 00070e56-ecee0009-5cc18a16-1ac28ea0@INTERNALIPOFPHONE
        CSeq: 102 BYE
        User-Agent: FPBX-2.11.0(11.16.0)
        Proxy-Authorization: Digest username="867", realm="asterisk", algorithm=MD5, uri="sip:ASTERISKREALM", nonce="3d3wef1e3efwce", response="5abae01f3dfe50aaefwefddw93919cf791eeae5f"
        X-Asterisk-HangupCause: Normal Clearing
        X-Asterisk-HangupCauseCode: 16
        Content-Length: 0

Here is the 404 from the Cisco:
Session Initiation Protocol (404)
    Status-Line: SIP/2.0 404 Not Found
    Message Header
        Via: SIP/2.0/UDP KAMAILIO:5060;branch=z9hG4bK1d8c.d024c8e7c11b4aa463be0af25e9015cc.0,SIP/2.0/UDP FREESWITCH:11000;received=FREESWITCH;rport=11000;branch=z9hG4bKB197XjSKQB1BS
        From: <sip:1237@REALM>;tag=9Sa02DmUgvBFS
        To: "867" <sip:867@REALM>;tag=00070e56ecee0009207708af-1d4c4d03
        Call-ID: 00070e56-ecee0004-79a4257f-2b6ba599@INTERNALIPOFPHONE
        Date: Fri, 16 Sep 2016 02:58:16 GMT
        CSeq: 96661284 BYE
        Content-Length: 0



Alexander Mustafin

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Sep 23, 2016, 12:00:34 AM9/23/16
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Hi, Brandon.

Try to change NAT setting on the phone, and maybe set external NAT address. I have both 7940/60 working fine.

Gordon Sims

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Sep 27, 2016, 7:15:07 AM9/27/16
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Alexander,

Were you able to get DNS SRV records to work with the Cisco Phones?  if so, what is the settings on it that you used in the xml file? or is there something special that needs to be setup on the DNS Server side?  On the current setup, I can get it to work with the Aastra, Polycoms and the newer Cisco phones like the SPA's, but not the 79XX or 99XX.
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