Re: [Serusers] NAT Related Voicemail Timeout Problem (HELP!)

2 Sep 2005

Hi Corey,
A few comments:
- It seems that after the timeout, a new INVITE is generated to the UA, 
right before the forwarding to Asterisk. Is this just noise or part of the 
actual call you documented?
- The record routing and path of the messages look fine AFAICS, but you 
don't mention one crucial point: Does the caller receive the OK or not? You 
assume it is a NAT problem, but are you sure? Maybe the caller discards the 
message for some reason?
- You haven't shown the initial INVITE from .7. Does it originate on port 
5060?  Some GWs will be asynchronous and expect reply on the high UDP port 
used for the INVITE. Your NAT detection may then detect the GW as NATed and 
rewrite, which may cause a problem.
- I remember a problem where Asterisk was involved where ;lr=on caused a 
problem. I cannot remember if it is relevant, but you may want to search in 
the archives
- A log from the caller would tell you if the OK is received or dropped

g-)

----- Original Message ----- 
From: "Corey S. McFadden" &lt;csm-lists(a)csma.biz&gt;
To: &lt;serusers(a)lists.iptel.org&gt;
Sent: Thursday, September 01, 2005 08:20 PM
Subject: [Serusers] NAT Related Voicemail Timeout Problem (HELP!)

...

 We're working through an annoying problem with voicemail and the
 fr_inv_timer and after banging our heads against it for a while figured
 we'd reach out for a little help.  It's probably something obvious, but
 here's the scenario:

 1. SIP DID rings SER
 2. SER Routes to registered UA (UA behind NAT)
 3. UA Rings
 4. fr_inv_timer is hit and failure_route[1] is invoked
 5. failure_route[1] rewrites host to send call to Asterisk for VM
 6. Asterisk gets the call and starts processing but the calling party
 never gets the OK, gets no audio, and dies with a fast busy.

 When the UA is NOT registered or registered without going through NAT
 there isn't a problem.  Asterisk OK's and works fine.

 This is probably some moronic NAT issue but we're not sure how to proceed.
 I'm going to follow with a 'sip debug' message from Asterisk, some ngrep
 from the SER machine (with commentary), and finally the SER cnf itself.

 If anyone can help we would be incredibly grateful! (Seriously!)

 Thanks,
 -Corey

 ** IP Legend
 xxx.xxx.xxx.7 - SIP Originating Source
 xxx.xxx.xxx.36 - SER
 xxx.xxx.xxx.80 - Asterisk VM Server
 216.NAT.CLI.207 - NAT UA Public IP
 192.168.249.83 - Private IP of UA

 5410000000 - DID & UA ID
 6100000000 - Outside Caller ID

 ** Versions
 SER: 0.9.4-rc3
 rtpproxy: 20040107 Extension 20050322

 ** Asterisk SIP Debug shows this on the second retry:

 Retransmitting #1 (no NAT) to xxx.xxx.xxx.36:5060:
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bK9e09.116856a3.2
 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-5Jn7520l3PhAH6h-0
 Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=5Jn7520l3PhAH6h-IPTrunk-81-17-23atxxx.xxx.xxx.7;lr=on>
 From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=5Jn7520l3PhAH6h-IPTrunk-81-17-23atxxx.xxx.xxx.7
 To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as625cebc0
 Call-ID: 5Jn7520l3PhAH6h(a)xxx.xxx.xxx.7
 CSeq: 101 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
 Contact: <sip:5410000000@xxx.xxx.xxx.80>
 Content-Type: application/sdp
 Content-Length: 237

 v=0
 o=root 29275 29275 IN IP4 xxx.xxx.xxx.80
 s=session
 c=IN IP4 xxx.xxx.xxx.80
 t=0 0
 m=audio 11476 RTP/AVP 0 3 8 97
 a=rtpmap:0 PCMU/8000
 a=rtpmap:3 GSM/8000
 a=rtpmap:8 PCMA/8000
 a=rtpmap:97 iLBC/8000
 a=silenceSupp:off - - - -

 After re-transmitting one more time it will finally fail.

 *** NGREP OUTPUT (from SER box)

 #
 U xxx.xxx.xxx.7:5060 -> xxx.xxx.xxx.36:5060
  INVITE sip:5410000000@xxx.xxx.xxx.36:5060 SIP/2.0..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@192.168.249.83:5060 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23a
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at
  xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route:
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at
  xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  ....

 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  CANCEL sip:5410000000@192.168.249.83:5060 SIP/2.0..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.0..From: sip
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  ACK sip:5410000000@192.168.249.83:5060 SIP/2.0..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at
  xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 ...
 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
 ...
 # (ASTERISK SERVER ENTERS THE PICTURE HERE)
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.80:5060
  INVITE sip:5410000000@xxx.xxx.xxx.80 SIP/2.0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at209.
  216.177.7;lr=on>..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;rport=
  5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Max-Forwards: 5..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-I
  PTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 I
  NVITE..Remote-Party-Id:

<sip:6100000000@xxx.xxx.xxx.7:5060>;party=calling;privacy=off;id-type=subscriber..Expires:
 180..All
  ow: INVITE,CANCEL,BYE,ACK..Contact: sip:xxx.xxx.xxx.7:5060..User-Agent:
 Tekelec-7000/r4.0..Content-Type: application/sdp..C
  ontent-Length: 169....v=0..o=- 1 1 IN IP4 xxx.xxx.xxx.7..s= ..c=IN IP4
 xxx.xxx.xxx.36..t=0 0..m=audio 35314 RTP/AVP 0..a=pt
  ime:20..a=rtpmap:0 PCMU/8000..a=direction:active..a=nortpproxy:yes..

 #
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  CANCEL sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Via: SIP/2.0/UDP
 ...
 #  (ASTERISK SERVER RESPONDS)
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;bran
  ch=z9hG4bK-00101-kYn356Vl3PhGzel-0..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23at209.216.1
  77.7..To: sip:5410000000@xxx.xxx.xxx.36:5060..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asteris
  k PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Length: 0
  ....

 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 ...
 #
 U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP
 ...
 # (SER ACKs the UAs' 487)
 U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060
  ACK sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Via: SIP/2.0/UDP
 ...
 # (ASTERISK SENDS THE OK)
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z
  9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o
  n>..From:

sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@209.216
  .160.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101
 INVITE..User-Agent: Asterisk PBX..Allow: INV
  ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con
  tent-Length: 237....v=0..o=root 29275 29275 IN IP4
 xxx.xxx.xxx.80..s=session..c=IN
 IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184
   RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of
  f - - - -..

 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:209.216.160.
  36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel
  -IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel(a)209.216.1
  77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
 CANCEL,
 OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54
  14362504@xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length:
 255....v=0..o=root 29275 29275 IN IP4 209.216.160.
  80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 
 8
 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a
  =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - -
 -..a=nortpproxy:yes..
 # (ASTERISK SENDS ANOTHER OK)
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z
  9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o
  n>..From:

sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@209.216
  .160.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101
 INVITE..User-Agent: Asterisk PBX..Allow: INV
  ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con
  tent-Length: 237....v=0..o=root 29275 29275 IN IP4
 xxx.xxx.xxx.80..s=session..c=IN
 IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184
   RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of
  f - - - -..

 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:209.216.160.
  36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel
  -IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel(a)209.216.1
  77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
 CANCEL,
 OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54
  14362504@xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length:
 255....v=0..o=root 29275 29275 IN IP4 209.216.160.
  80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 
 8
 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a
  =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - -
 -..a=nortpproxy:yes..
 # (ASTERISK SENDS YET ANOTHER OK)
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z
  9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o
  n>..From:

sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@209.216
  .160.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101
 INVITE..User-Agent: Asterisk PBX..Allow: INV
  ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con
  tent-Length: 237....v=0..o=root 29275 29275 IN IP4
 xxx.xxx.xxx.80..s=session..c=IN
 IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184
   RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of
  f - - - -..

 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:209.216.160.
  36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel
  -IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel(a)209.216.1
  77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
 CANCEL,
 OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54
  14362504@xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length:
 255....v=0..o=root 29275 29275 IN IP4 209.216.160.
  80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 
 8
 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a
  =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - -
 -..a=nortpproxy:yes..
 # (ASTERISK SENDS ANOTHER OK)
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z
  9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o
  n>..From:

sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@209.216
  .160.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101
 INVITE..User-Agent: Asterisk PBX..Allow: INV
  ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con
  tent-Length: 237....v=0..o=root 29275 29275 IN IP4
 xxx.xxx.xxx.80..s=session..c=IN
 IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184
   RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of
  f - - - -..

 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:209.216.160.
  36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel
  -IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel(a)209.216.1
  77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
 CANCEL,
 OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54
  14362504@xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length:
 237....v=0..o=root 29275 29275 IN IP4 209.216.160.
  80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 
 8
 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a
  =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -..

 # ASTERISK KEEPS TRYING TO SEND OK
 U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z
  9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o
  n>..From:

sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@209.216
  .160.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101
 INVITE..User-Agent: Asterisk PBX..Allow: INV
  ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact:
 <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con
  tent-Length: 237....v=0..o=root 29275 29275 IN IP4
 xxx.xxx.xxx.80..s=session..c=IN
 IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184
   RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8
 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of
  f - - - -..

 #
 U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP
 xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route:
 <sip:209.216.160.
  36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From:
 sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel
  -IPTrunk-83-17-23atxxx.xxx.xxx.7..To:
 sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID:
 kYn356Vl3PhGzel(a)209.216.1
  77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK,
 CANCEL,
 OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54
  14362504@xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length:
 237....v=0..o=root 29275 29275 IN IP4 209.216.160.
  80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 
 8
 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a
  =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -..

 ** At this point the call is in fast-busy and Asterisk shows a failure.

 **** SER CFG
 # ser.cfg
 ############################################################

 debug=3
 fork=yes
 log_stderror=yes

 listen=xxx.xxx.xxx.36
 port=5060
 children=4
 alias=whatever.net
 alias=whatever.whatever.net

 dns=no
 rev_dns=no

 fifo="/tmp/ser_fifo"
 fifo_db_url="mysql://ser:serpass@localhost/ser"

 ###
 # Modules
 ###

 loadmodule "/usr/local/lib/ser/modules/mysql.so"
 loadmodule "/usr/local/lib/ser/modules/sl.so"
 loadmodule "/usr/local/lib/ser/modules/tm.so"
 loadmodule "/usr/local/lib/ser/modules/acc.so"
 loadmodule "/usr/local/lib/ser/modules/rr.so"
 loadmodule "/usr/local/lib/ser/modules/exec.so"
 loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
 loadmodule "/usr/local/lib/ser/modules/usrloc.so"
 loadmodule "/usr/local/lib/ser/modules/registrar.so"
 loadmodule "/usr/local/lib/ser/modules/auth.so"
 loadmodule "/usr/local/lib/ser/modules/auth_db.so"
 loadmodule "/usr/local/lib/ser/modules/nathelper.so"
 loadmodule "/usr/local/lib/ser/modules/textops.so"
 loadmodule "/usr/local/lib/ser/modules/uri_db.so"
 loadmodule "/usr/local/lib/ser/modules/uri.so"
 loadmodule "/usr/local/lib/ser/modules/permissions.so"

 ###
 # Module Paramaters
 ###

 modparam("auth_db|uri_db|usrloc", "db_url",
 "mysql://ser:serpass@localhost/ser")
 modparam("auth_db", "calculate_ha1", 1)
 modparam("auth_db", "password_column", "password")

 modparam("nathelper", "natping_interval", 30)
 modparam("nathelper", "ping_nated_only", 1)
 modparam("nathelper", "rtpproxy_sock",
"unix:/var/run/rtpproxy.sock")

 # VM Timeout
 modparam("tm", "fr_inv_timer", 20)

 modparam("usrloc", "db_mode", 2)
 modparam("permissions", "db_mode", 0)
 modparam("permissions", "trusted_table", "trusted")
 modparam("registrar", "nat_flag", 6)
 modparam("rr", "enable_full_lr", 1)
 modparam("exec", "setvars", 1)

 modparam("acc", "log_level", 1)
 modparam("acc", "log_flag", 1)
 modparam("acc", "db_url",
"mysql://ser:serpass@localhost/ser")
 modparam("acc", "db_flag", 1)

 ###
 # Routing Section
 ###

 route {

        # Basic Checks
        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483", "Too Many Hops");
                break;
        };
        if (msg:len > max_len) {
                sl_send_reply("513", "Message Overflow");
                break;
        };
        if (method!="REGISTER") {
                record_route();
        };

        if (method=="BYE" || method=="CANCEL") {
                unforce_rtp_proxy();
        }

        # Loose Routing
        if (loose_route()) {
                route(1);
                break;
        };

        # Call Processing
        if (uri!=myself) {
                route(1);
                break;
        };
        if (uri==myself) {

                if (method=="CANCEL") {
                        route(3);
                        break;
                } else if (method=="INVITE") {
                        route(3);
                        break;
                } else  if (method=="REGISTER") {
                        route(2);
                        break;
                };

                lookup("aliases");
                if (uri!=myself) {
                        route(1);
                        break;
                };

                if (!lookup("location")) {
                        sl_send_reply("404", "User Not Found");
                        break;
                };
        };

        route(1);
 }

 route[1] {

        ###
        # Default Call Handling
        ###

        # NAT Fix
        if (isflagset(6) && !isflagset(9)) {
                fix_nated_sdp("1");
                force_rport();
                fix_nated_contact();
                force_rtp_proxy();
                setflag(9);
        };

        # On call timeout to voicemail
        t_on_failure("1");
        t_on_reply("1");

        # Log
        setflag(1);

        if (!t_relay()) {
                if (method=="INVITE" && isflagset(6)) {
                  unforce_rtp_proxy();
                };
                sl_reply_error();
        };
 }

 route[2] {

        ###
        # REGISTER method
        ###

        if (!search("^Contact:\ +\*") && nat_uac_test("19"))
{
                setflag(6);
                fix_nated_register();
                force_rport();
        };

        sl_send_reply("100", "Trying");

        if (!www_authorize("whatever.net","subscriber")) {
                www_challenge("whatever.net","0");
                break;
        };

        if (!check_to()) {
                sl_send_reply("401", "Unauthorized");
                break;
        };

        consume_credentials();

        if (!save("location")) {
                sl_reply_error();
        };
 }

 route[3] {

        ###
        # INVITE & CANCEL
        # URI matching is here.
        ###

        t_on_failure("1");

        if (nat_uac_test("19")) {
                log(1," INVITE: NAT Detected. Setting flag 6.\n");
                setflag(6);
        }

        if (method=="INVITE" && src_ip==xxx.xxx.xxx.7) {
                !allow_trusted()) {

                if (!proxy_authorize("whatever.net","subscriber")) {
                       proxy_challenge("whatever.net","0");
                       break;
                } else if (!check_from()) {
                       sl_send_reply("403", "Use From=ID");
                       break;
                };

                consume_credentials();
        };

        lookup("aliases");
 #        if (uri!=myself) {
 #               force_rtp_proxy();
 #                route(1);
 #                break;
 #        };

        if (!lookup("location")) {

                if (uri=~"^sip:911@") { # Pass 911 calls first
                        route(5);
                        break;
                };

                if (uri=~"^sip:[0-9]{7}@") { # EXPAND 7-DIGIT CALL
                        exec_dset("/root/7digit.pl");
                };
                # Try location lookup again

                if (!lookup("location")) {

                        if (uri=~"^sip:[0-9]{10}@") {
                                # DB Lookup of NPANXX route
                                exec_dset("/root/npa_nxx.pl");
                                route(1);
                                break;
                        };

                        if (uri=~"^sip:1[0-9]{10}@") {
                                # DB Lookup of NPANXX route
                                exec_dset("/root/npa_nxx.pl");
                                route(1);
                                break;
                        };
                        # Call error message
                        route(7);

                        break;
                };
        };

 #       if (isflagset(6)) {
 #               force_rport();
 #               fix_nated_contact();
 #               force_rtp_proxy();
 #       };

        t_on_reply("1");

        if (!t_relay()) {
                if(isflagset(6)) {
                        unforce_rtp_proxy();
                }
                sl_reply_error();
        };
 }

 route[4] {

        ###
        # Primary LD Gateway
        ###

        rewritehost("xxxxxxxxxxxxxxxx");
        route(1);
 }

 route[5] {

        ###
        # Local PSTN Gateway
        ###
        rewritehost("xxx.xxx.xxx.7");
        route(1);
 }

 route[6] {

        ###
        # Calls to VM
        ###
        rewritehost("xxx.xxx.xxx.80");
        route(1);
 }

 route[7] {

        ###
        # Call Failure
        ###
        rewriteuri("sip:7110000001@xxx.xxx.xxx.80");
        route(1);
 }

 failure_route[1] {

        ###
        # Voicemail
        ###

        log(1,"Failure Route Hit\n");
        rewritehost("xxx.xxx.xxx.80");
        append_branch();
        route(1);
 }

 onreply_route[1] {

        if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") {
                if (!search("^Content-Length:\ +0")) {
                        force_rtp_proxy();
                };
        };

        if (nat_uac_test("1") && !isflagset(9)) {
                fix_nated_contact();
        };
 }

 --
 Corey S. McFadden (c(a)csma.biz)
 McFadden Associates - Technology Consultants
 phone 215-825-2121 x510  - web.csma.biz

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Re: [Serusers] NAT Related Voicemail Timeout Problem (HELP!)