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rogelio.perez at gmail... Guest
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Posted: Fri Oct 16, 2009 1:56 am Post subject: [Freeswitch-users] SIP CANCEL error |
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Hi guys,
I'm having a signaling issue on FS when calling from an endpoint to a
GW.
To reproduce the error I just call and hungup the call right away.
FS gets the CANCEL and replies back with 481 Call/Transaction Does Not
Exist, so the call continues and eventually connects, even when the A
side of the call is closed.
I have verified that the Call-ID of the CANCEL matches that from the
INVITE, so I see no reasons for FS to reject it.
SIP+FS logs in pastebin 10738.
Any ideas?
Thanks,
Rogelio
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anthony.minessale at g... Guest
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Posted: Sat Oct 17, 2009 11:57 am Post subject: [Freeswitch-users] SIP CANCEL error |
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I think the phone is making a mistake with the CANCEL packet.
Here is the Via header from the INVITE the phone sends:
Via: SIP/2.0/UDP 190.137.190.131;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792
Here is the Via header from the CANCEL
Via: SIP/2.0/UDP 10.0.0.8;rport;branch=z9hG4bKac109601000000fb4ad814425106977400000792
it's identical aside from the private address contained in the CANCEL and a public one in the INVITE
So the anti-nat settings on your phone are not working properly and they forgot to substitute the public IP for the CANCEL.
That makes Sofia unable to match it to an existing call.
On Fri, Oct 16, 2009 at 1:44 AM, Rogelio Perez <rogelio.perez@gmail.com (rogelio.perez@gmail.com)> wrote:
--
Anthony Minessale II
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AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
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IRC: irc.freenode.net #freeswitch
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sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
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