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Summary:ASTERISK-14814: Interoperability with Exchange 2007 UM
Reporter:rsw686 (rsw686)Labels:
Date Opened:2009-09-10 16:38:18Date Closed:2009-10-08 17:35:39
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk-exchange-1.6.1.4.log
( 1) asterisk-exchange-1.6.1.5.log
( 2) asterisk-exchange-1.6.1.5-port-5067.log
Description:Starting with versions 1.6.0.14 and 1.6.1.5 Asterisk fails to connect with Exchange 2007 UM. Both versions 1.6.0.13 and 1.6.1.4 which were released on 18-Aug-2009 work.

****** ADDITIONAL INFORMATION ******

I have installed Asterisk on CentOS 5.3 with the following commands.

./configure
make
make install
make config
make samples

Then I replaced sip.conf and extensions.conf with the the below.

SIP.CONF
[general]
context = default
bindport = 5060
bindaddr = 0.0.0.0
tcpbindaddr = 0.0.0.0
tcpenable = yes
promiscredir = yes

[SIP_VM]
type = peer
host = 10.9.1.13
qualify = yes
transport = tcp

[8678]
type = friend
callerid = Exchange User <8678>
secret = 1234
host = dynamic
canreinvite = no
dtmfmode = rfc2833
mailbox = 8678
disallow = all
allow = ulaw
transport = udp

EXTENSIONS.CONF
[general]
static=yes
writeprotect=no

[globals]

[default]
exten => 3300,1,Answer
exten => 3300,n,Dial(SIP/SIP_VM/${EXTEN})
exten => 3300,n,Busy
Comments:By: rsw686 (rsw686) 2009-09-10 18:43:01

I wanted to provide some logs from the Asterisk console. On version 1.6.1.4 when a call goes to Exchange I receive the following output.

 == Using SIP RTP CoS mark 5
   -- Executing [3300@default:1] Answer("SIP/8678-094863c0", "") in new stack
   -- Executing [3300@default:2] Dial("SIP/8678-094863c0", "SIP/SIP_VM/3300") in new stack
 == Using SIP RTP CoS mark 5
   -- Called SIP_VM/3300
   -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
   -- Now forwarding SIP/8678-094863c0 to 'SIP/3300::::TCP@10.9.1.13:5065' (thanks to SIP/SIP_VM-094f1cd0)
 == Using SIP RTP CoS mark 5
   -- SIP/10.9.1.13:5065-094f6f58 is ringing
   -- SIP/10.9.1.13:5065-094f6f58 answered SIP/8678-094863c0
   -- Packet2Packet bridging SIP/8678-094863c0 and SIP/10.9.1.13:5065-094f6f58

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
8678/8678                  10.9.5.107       D          32971    Unmonitored
SIP_VM                     10.9.1.13                   5060     OK (1 ms)

On version 1.6.1.5 when trying to call Exchange I see

 == Using SIP RTP CoS mark 5
   -- Executing [3300@default:1] Answer("SIP/8678-09978b88", "") in new stack
   -- Executing [3300@default:2] Dial("SIP/8678-09978b88", "SIP/SIP_VM/3300") in new stack
 == Using SIP RTP CoS mark 5
[Sep 10 19:37:53] WARNING[12237]: app_dial.c:1528 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing [3300@default:3] Busy("SIP/8678-09978b88", "") in new stack

asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
8678/8678                  10.9.5.107       D          32971    Unmonitored
SIP_VM                     10.9.1.13                   5060     UNREACHABLE

This is coming from the same machine. All I have done is stop asterisk, run make install on the other version, and then started asterisk.
By: Russell Bryant (russell) 2009-09-10 21:49:49

Can you try the latest code from svn and see if it has been fixed?  There have been some related changes in the past week or two.

$ svn co http://svn.digium.com/svn/asterisk/branches/1.6.0

or

$ svn co http://svn.digium.com/svn/asterisk/branches/1.6.1

By: rsw686 (rsw686) 2009-09-10 22:21:27

I just checked out the 1.6.1 branch, but it didn't fix the problem. I am getting the same output as with 1.6.1.5. I reinstalled 1.6.1.4 and it works again.

I was looking through http://svn.asterisk.org/svn/ but couldn't find a way to see the revisions on individual files. I'm assuming the issue is with either chan_sip.c or rtp.c.
By: Russell Bryant (russell) 2009-09-10 22:37:07

Thanks for the feedback.

Try this link for looking around: http://svn.asterisk.org/view
By: Russell Bryant (russell) 2009-09-10 22:37:53

Another useful item to include would be SIP traces for both the working and non-working cases.
By: rsw686 (rsw686) 2009-09-14 09:12:55

I turned off the qualify setting so Asterisk would attempt to make the connection. I've attached three log files: 1.6.1.4, 1.6.1.5, and 1.6.1.5 with the port set to 5067 which Exchange was requesting a redirect for. For each test I started up Asterisk, opened X-Lite, dialed Exchange (3300), turned on sip debugging, and dialed Exchange (3300). It looks like both the UDP/TCP bridge and the ability to redirect with a Moved Temporarily response is broken in 1.6.1.5.
By: rsw686 (rsw686) 2009-09-14 09:19:52

Just noticed an email on the list about bug / issue 15826. This is the same issue that I am having.
By: rsw686 (rsw686) 2009-09-22 08:56:36

Is there any resolution for this as I see that 1.6.1.7 is coming out soon?
By: rsw686 (rsw686) 2009-09-22 14:18:21

I just installed 1.6.1.7-rc1 and can confirm that the problem has been resolved.

Connected to Asterisk 1.6.1.7-rc1 currently running on asterisk (pid = 20860)
asterisk*CLI> core set verbose 10
Verbosity was 0 and is now 10
asterisk*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
8678/8678                10.9.5.107       D          60102    OK (101 ms)
SIP_VM                     10.9.1.13                   5060     OK (1 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
 == Using SIP RTP CoS mark 5
   -- Executing [3300@default:1] Answer("SIP/8678-0997bf68", "") in new stack
   -- Executing [3300@default:2] Dial("SIP/8678-0997bf68", "SIP/SIP_VM/3300") in new stack
 == Using SIP RTP CoS mark 5
   -- Called SIP_VM/3300
   -- Got SIP response 302 "Moved Temporarily" back from 10.9.1.13
   -- Now forwarding SIP/8678-0997bf68 to 'SIP/3300::::TCP@10.9.1.13:5065' (thanks to SIP/SIP_VM-0997af08)
 == Using SIP RTP CoS mark 5
   -- SIP/10.9.1.13:5065-09979878 is ringing
   -- SIP/10.9.1.13:5065-09979878 answered SIP/8678-0997bf68
 == Spawn extension (default, 3300, 2) exited non-zero on 'SIP/8678-0997bf68'


By: Elazar Broad (ebroad) 2009-10-08 17:35:39

As the reported states, the issue seems to be resolved in 1.6.1.7-rc1.