Answering my own question as I finally sat down and invested some time in it....
After a few digging here and there in a number of traces I ran, and I noticed
an anomaly [I just like the word better than bug ] in the SIP VIA Header coming
from SIPX specially at the Request/INVITE section.
Long story short, {full log trace available for the asking...}
- In a failed call, SIPX is sending an empty "received" parameter to Exchange,
thus unable to respond back which then causes a 480 timeout to occur
Via: SIP/2.0/TCP
192.168.2.23;branch=z9hG4bK-adbe851de56f0b748ed6a3125fe8c49d;receivedTransport:
TCPSent-by Address: 192.168.2.23Branch:
z9hG4bK-adbe851de56f0b748ed6a3125fe8c49dReceived: received
- However, you would notice below the received field is missing all together on
a successful as you can see here
Via: SIP/2.0/TCP
192.168.2.23;branch=z9hG4bK-580ad10bfa826656777c119dd27aeb71Transport:
TCPSent-by Address: 192.168.2.23Branch: z9hG4bK-580ad10bfa826656777c119dd27aeb71
Apparently, according to RFC3486 such VIA Header should either drop the
"received" field or when referenced, it should be well formatter, i.e.
received=0.0.0.0 whereas 0.0.0.0 is a valid IP address.
After further digging, this happened to be a known anomaly within the SIPX
world which is then resolved with version 3.8 and above.For reference, it's
found here.
http://track.sipfoundry.org/browse/XPR-66?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
Thanks to all for your input on and off list; it's been much appreciated!
Claudius
> From: [EMAIL PROTECTED]> To: [email protected]> Date: Mon, 30 Jul 2007 18:21:26
> -0400> Subject: [on-asterisk] Asterisk with Exchange Srv 2007 UM> > Hi
> everyone,> > I believe to have seen something along these lines long time
> ago. I personally have been able to integrate Asterisk with Exchange Server
> 2007 and divert Asterisk VoiceMail to Exchange......and everything has been
> working just fine except I get a lot of time out from time to time: 2/5 calls
> to Exchange UM causes a "480 Request time out"; thus asterisk will keep
> ringing as if it was calling a number that doesn't exist and eventually throw
> a congestion tone or disconnect.> > Tech. specs> ------------> SIP Gateway:
> Asterisk 1.2.18> SIP Proxy: SIPx 3.6> VoiceMail: Exchange Server 2007> > The
> call route is as follow:DID >> Asterisk >> ring local *extension [if
> unavail.]>> VM [Exchange Srv]> > Communications between Asterisk and Exchange
> pass thru SIPx> > Has anyone been successful integrating Exchange UM as a
> VoiceMail system for Asterisk? And if so, anything would you suggest I look
> into in t-shooting efforts?> > Thanks and have all a great one!> > Claudius>
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