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Call immediately hung up.

Oct 25, 2012 at 1:35 PM
Edited Oct 25, 2012 at 1:38 PM



I have been checking and working out the SoftPhone code for a while and I managed to get things working pretty much. 


I can make calls and have conversation. I call also be called but the moment I accept the call the call will be hung up. Do you have any idea what goes wrong?

My SIP server gives the following message:


25-10-2012 14:30:22.627 L:32.1[Extn:200] failed to reach Extn:100, reason User Requested

25-10-2012 14:30:22.627 Call to T:Extn:100@[Dev:sip:100@] from L:32.1[Extn:200] failed, cause: Cause: 200 Ok/INVITE from

25-10-2012 14:30:22.627 [CM503003]: Call(C:32): Call to <sip:100@> has failed; Cause: 200 Ok/INVITE from

25-10-2012 14:30:21.037 [CM503025]: Call(C:32): Calling T:Extn:100@[Dev:sip:100@] for L:32.1[Extn:200]

25-10-2012 14:30:20.989 [CM503027]: Call(C:32): From: Extn:200 ("Kevin" <sip:200@>)  to  T:Extn:100@[Dev:sip:100@]

25-10-2012 14:30:20.989 [CM503004]: Call(C:32): Route 1: from L:32.1[Extn:200] to T:Extn:100@[Dev:sip:100@]

25-10-2012 14:30:20.987 [CM503001]: Call(C:32): Incoming call from Extn:200 to <sip:100@>


It says the call failed while the status code says 200? 

When putting a break point on SIPTransportRequestReceived it says I receive a BYE messages from the client I called.

The client I call is an actual working SIP phone program from 3CX.

Do you have any clue what's going on?

Oct 26, 2012 at 10:24 AM

It's a bit hard to say without seeing a full SIP trace but sometimes user agents will immediately hang up a call if they don't like the codec being used or perhaps they don't like something in the RTP packets. If you catch a full SIP trace with wireshark and post it up here that should provide more information to help you with.