It's definitely a bug on the phone. I would try setting all Aastra NAT related features off and see if it stops getting confused like this. Freeswitch should be able to handle NAT from the far end.
On 12/02/2011 08:18 PM, Charlie Orford wrote:Thanks Joegen, I hadn't picked up on that.
Is there any valid reason why the phone would swap to using it's internal IP (192.168.0.175) for the Via and Contact headers when responding to the auth challenge? Or does this look more like a bug with the phone?
On 02/12/2011 12:50, Joegen Baclor wrote:For some reason, the aastra phone is sending a different contact and via address after it gets challenged by freeswitch.
Original address was
Via: SIP/2.0/UDP 7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport (LINE# 35)
Then it changes to
Via: SIP/2.0/UDP 192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport (LINE# 187)
It then sends a CANCEL using Via: SIP/2.0/UDP 7.7.7.7:6060 (LINE# 961) which freeswitch isn't able to match to a transaction because it is probably expecting 192.168.0.175.
On 12/02/2011 07:19 PM, Charlie Orford wrote:Hi Anthony
We are using Aastra 57i phones with the latest firmware (v3.2.2.56 from June 2011).
However, it looks like a specific issue with the Aastra 57i as I have just replicated the call and cancel situation using a 3CX softphone and in this case the CANCEL request gets honoured by FS.
Here's a copy of a failed CANCEL test using the Aastra phone: http://pastebin.freeswitch.org/17922
Here's a copy of a successful CANCEL test using the 3CX softphone: http://pastebin.freeswitch.org/17923
sofia loglevel all 9 is on for both tests.
Charlie
On 01/12/2011 21:53, Anthony Minessale wrote:try same failed call test with sofia loglevel all 9
Also, try some other phone or device that does not have the problem, and create a cancel situation the same way and see if you can find a difference.
Finally, have you tried the latest firmware on the phone?
On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford <charlie.orford@attackplan.net> wrote:
Hi list,
When we make a call from an FS extension to a PSTN number (via our ITSP
gateway provider) and hang-up before the call completes, FS replies to
the CANCEL request with "481 Call/Transaction Does Not Exist" and the
call continues to ring on the remote end. If we hangup after the call
has connected, it works with no problem.
I have looked through a SIP trace of this happening and to my (untrained
eye) nothing seems obviously wrong (i.e. tag, call id and branch values
all seem to be correct). I'm using the latest git snapshot from
2011-11-30 18-14-24 -0600.
For a sip trace showing the problem, see:
http://pastebin.freeswitch.org/17908
For a sip trace showing a successful hangup, see:
http://pastebin.freeswitch.org/17909
*Note: ip addresses, domain and called number have been altered for privacy.
Any help or insight is much appreciated.
Kind Regards,
Charlie
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