Problem in BYE send from Webrtc2sip (Version 2.6.0)

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What steps will reproduce the problem?
1. Make a Call from Sip Client (softphone registered on a PBX)
2. Call is Answer in SIPML5, next Hangup the Call in SipML5
3. Caller (Softphone) not drop the call because an Error in BYE

What is the expected output? What do you see instead?

Softphone and Sipml5 drops the call.

What version of the product are you using? On what operating system?

Webrtc2sip VERSION: 2.6.0 in Centos 6.4

Please provide server logs with DEBUG level equal to INFO

fragment of Wireshark capture:

1 Request: BYE 
sip:XXX.XXX.140.140:5061;transport=tcp;gsid=838d6170-e2e0-11e4-a45f-b4b52f6a27d4
2 Status: 407 Proxy Authentication Required
3 Request: BYE 
sip:XXX.XXX.140.140:5061;transport=tcp;gsid=838d6170-e2e0-11e4-a45f-b4b52f6a27d4
4 Status: 403 Forbidden (Unauthorized)


Detail of Proxy-Authorization Header in Bye send from Webrtc2sip (line 3), 
error in Username, "webrtc2sip" in not my username, username is 11001:


Digest 
username=\"webrtc2sip\",realm=\"mypbx.com\",nonce=\"14cb9826a095c3d82caa2ffbfc40
8dda3723ac56fac\",uri=\"sip:10.252.140.140:5061;transport=tcp;gsid=838d6170-e2e0
-11e4-a45f-b4b52f6a27d4\",response=\"9bc020fdc826723d052639974dc30ff5\",algorith
m=MD5,cnonce=\"0d6d6eb700ef0bdc2c71adb5e228144d\",opaque=\"1234567890abcedef\",q
op=auth,nc=00000001

Original issue reported on code.google.com by [email protected] on 14 Apr 2015 at 8:08

Attachments:

• capture.jpg

[GoogleCodeExporter]

Call Flow.

P.D.- Excelent Work!!!! I Like much webrtc2sip.

Original comment by [email protected] on 14 Apr 2015 at 8:19

Attachments:

• capture2.jpg