Problem in BYE send from Webrtc2sip (Version 2.6.0)
GoogleCodeExporter opened this issue · comments
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 lperezu1...@gmail.com on 14 Apr 2015 at 8:08
Attachments:
Call Flow.
P.D.- Excelent Work!!!! I Like much webrtc2sip.
Original comment by lperezu1...@gmail.com on 14 Apr 2015 at 8:19
Attachments: