Thanks Anthony, We have turned off all nat-related anything, but are still having an issue. However, here is more information: -- We are running FreeSwitch on Windows Server 2003 R2 -- We are running FreeSwitch in a console app for testing Calls from Avaya are answered in the dial plan and a javascript is executed: <extension name="5060"> <condition field="destination_number" expression="^5060$"> <action application="javascript" data="Answer.js 5061 welcome.wav"/> </condition> </extension> Here's the js: var sAudioFilePath = "/sounds/"; session.answer(); if(argv.length > 1) { for (var i = 1; i < argv.length; i ++) { session.streamFile(sAudioFilePath + argv[i]); } } session.execute("transfer", argv[0]); exit(); After playing the file, the caller is transferred into a park, using this dialplan code: <extension name="5060_Park"> <condition field="destination_number" expression="^5061$"> <action application="set" data="fifo_music=$${sound_prefix}/sounds/pleasewait.wav"/> <action application="fifo" data="5061@$${domain} in"/> <action application="set" data="hangup_after_bridge=true"/> </condition> </extension> Here's the issue: If we hang up the call while it is parked, we see NOTHING in the sofia debug or freeswitch log. (console window). We deceided to log freeswitch log info to a file. If we shut down FreeSwitch when it is the state with a call that is parked, we do see the SIP BYE message, and everythig shuts down. It's as if some thread is hung. All of this only happens with an inbound Avaya trunk. If we dial from an anonymous SIP connection, everything works fine. Ideas? Regards, Gerry On Fri, Oct 17, 2008 at 3:30 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
make sure you have all of the nat related params off in FS just in case. edit your sip profile and comment out anything that says nat
On Fri, Oct 17, 2008 at 12:51 PM, Gerry Hull <gerry@pstn2.net> wrote:
I've solved most of my issues by trunking Freeswitch with Avaya SES. It was more on the SES configuration side. FreeSwitch seems to perform flawlessly.
Now the only problem is, if I dial into Freeswitch from an Avaya extension, and hangup the call, I get no SIP BYE or CANCEL from SES!
Oh joy!
Thanks for all your help on this, guys.
Gerry
On Thu, Oct 16, 2008 at 1:48 AM, Gayatri Kulkarni <xtpl.gayatri@gmail.com
wrote:
Gerry, Did you manage to take the ethereal trace? It would be definitely of more help and we can narrow down the actual problem Do you have access to Avaya SES?
To take the ethreal trace, you should: *1)telnet* <user>@<Avaya SES> //reduces size of the trace, ssh size usually goes in GBs 2)login 3)tethereal -i eth0 -f <path/to/filename> 4)try to make the call (get the 407) 5)kill tethereal (Ctrl+C) 6) copy the file to a place where you can sit and analyze it
You can analyze the trace yourself if you have wireshark installed, or send it over
-- Regards, Gayatri Kulkarni
On Wed, Oct 15, 2008 at 7:12 PM, Gerry Hull <gerry@pstn2.net> wrote:
Gayatri,
Any idea on how to enable this response in Freeswitch?
David,
Not sure of the "lr"...
On Wed, Oct 15, 2008 at 4:38 AM, Gayatri Kulkarni < xtpl.gayatri@gmail.com> wrote:
Thanks David! Gerry, From the debug info you have sent, looks like Avaya SES asks for PAI i.e Proxy Authentication Indication - It's a kind of challenge response authentication. After it receives the user's digest in response to this request (again), it authenticates the user. This is the normal behavior of Avaya SES. the users' digest is not sent again it seems!
On Wed, Oct 15, 2008 at 1:52 PM, David Knell <dave@3c.co.uk> wrote:
On Oct 15, 2008, at 9:00 AM, Gayatri Kulkarni wrote:
Record-Route: <sip:10.0.2.154:5060;lr>
Record-Route: <sip:10.0.2.151:5061;lr;transport=tls>
what's the 'lr' next to the port number?
short for 'loose routing' - see here for a bit of an explanation: http://www.tech-invite.com/Ti-sip-dialog.html
--Dave
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-- Regards, Gayatri Kulkarni
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