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sipml5
Automatically exported from code.google.com/p/sipml5
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call disconnection in api version SIPML5 API version = 1.5.230
#229
GoogleCodeExporter
opened
9 years ago
0
One way video, any browser, webrtc2sip and Asterisk to softphone
#228
GoogleCodeExporter
opened
9 years ago
2
Call.info() result is logged to console instead of given to user
#227
GoogleCodeExporter
closed
9 years ago
1
Temporary failure in name resolution
#226
GoogleCodeExporter
opened
9 years ago
0
Problem in BYE send from SIPML5 - Webrtc2sip (Version 2.6.0)
#225
GoogleCodeExporter
opened
9 years ago
0
Outbound call hangs up after 1 minute with FreeSwitch Session-Expire
#224
GoogleCodeExporter
opened
9 years ago
0
Parameters send to Event Listener Function
#223
GoogleCodeExporter
opened
9 years ago
2
No early audio in outgoing sipml call to an extension.
#222
GoogleCodeExporter
opened
9 years ago
2
sipml5 unable to play remote audio
#221
GoogleCodeExporter
opened
9 years ago
2
Available on main package managers
#220
GoogleCodeExporter
opened
9 years ago
0
Export to GitHub
#219
GoogleCodeExporter
opened
9 years ago
0
Video stream doesnt add in SipML5
#218
GoogleCodeExporter
opened
9 years ago
0
Firefox ICE resolution to incorrect 0.0.0.0 address
#217
GoogleCodeExporter
closed
9 years ago
2
Content length for SIP INFO packets is incorrect in latest release
#216
GoogleCodeExporter
closed
9 years ago
1
startBfcpShare not working
#215
GoogleCodeExporter
opened
9 years ago
0
Called in wrong state: STATE_INPROGRESS
#214
GoogleCodeExporter
opened
9 years ago
7
dtmf('#') and dtmf('*') are not detected correctly
#213
GoogleCodeExporter
opened
9 years ago
0
No Audio on Firefox 34+
#212
GoogleCodeExporter
opened
9 years ago
0
Sipml5 not connecting
#211
GoogleCodeExporter
closed
9 years ago
1
SUBSCRIBE does not contain route headers
#210
GoogleCodeExporter
opened
9 years ago
0
Double call "onGetUserMediaSuccess" and "createAnswer". Sipml5 can't answer on some calls and sends BYE.
#209
GoogleCodeExporter
opened
9 years ago
4
tsip_transac_layer.prototype.cancel_by_dialog incorrectly deletes transactions when there is more than one transaction
#208
GoogleCodeExporter
closed
9 years ago
2
Too large SIP INVITE
#207
GoogleCodeExporter
opened
9 years ago
0
dtmf formatted incorrectly
#206
GoogleCodeExporter
closed
9 years ago
3
enable_media_stream_cache setting does not work
#205
GoogleCodeExporter
opened
9 years ago
2
SIP INFO DTMF not compatible with Asterisk
#204
GoogleCodeExporter
closed
9 years ago
1
webrtc4all is not detected
#203
GoogleCodeExporter
opened
9 years ago
1
Bad invite send by client on Chrome 39 (linux 64)
#202
GoogleCodeExporter
closed
9 years ago
1
Call between chrome to softphone call automatically dropped
#201
GoogleCodeExporter
closed
9 years ago
1
Compatibility problem with Node-Webkit?
#200
GoogleCodeExporter
opened
9 years ago
0
Not working in safari 5.1.7
#199
GoogleCodeExporter
closed
9 years ago
1
Offer message containing c=IN IP4 0.0.0.0
#198
GoogleCodeExporter
closed
9 years ago
3
SipML5 + Asterisk : Remote host can't match request ACK
#197
GoogleCodeExporter
closed
9 years ago
4
Is 407 supported by sipml5
#196
GoogleCodeExporter
closed
9 years ago
2
WebRTC video call rejected asterisk11.130+sipml5
#195
GoogleCodeExporter
closed
9 years ago
1
which server can work with sipML5 api
#194
GoogleCodeExporter
opened
9 years ago
0
Race condition in hold/resume
#193
GoogleCodeExporter
opened
9 years ago
1
sip call from android uses loud speaker instead of front speaker.
#192
GoogleCodeExporter
opened
9 years ago
2
SIPml5 behind NAT, Freeswitch on Amazon EC2, Calll Hang Up
#191
GoogleCodeExporter
closed
9 years ago
2
Asterisk on AWS; RTP reciving at client end but no voice
#190
GoogleCodeExporter
opened
9 years ago
1
SUBSCRIBE request fails \failing to find dialog
#189
GoogleCodeExporter
opened
9 years ago
7
No video on Firefox 30 :onIceGatheringCompleted but no local sdp request is pending
#188
GoogleCodeExporter
opened
9 years ago
0
Settings ice_servers to [] does not disable ICE
#187
GoogleCodeExporter
opened
9 years ago
3
Incomming call without Video payload is refused
#186
GoogleCodeExporter
opened
9 years ago
0
UPD flows bidirectionally but one way audio on webphone
#185
GoogleCodeExporter
opened
9 years ago
3
Unable to Login to OfficeSip Server using chrome v. 32
#184
GoogleCodeExporter
opened
9 years ago
1
Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]
#183
GoogleCodeExporter
opened
9 years ago
6
Sipml5 SIP OPTIONS response SIP/2.0 405 Method Not Allowed
#182
GoogleCodeExporter
opened
9 years ago
0
Transfer call functionality is not working as expected in sipml5
#181
GoogleCodeExporter
opened
9 years ago
1
WebRTC Not working with Google Chrome 35.
#180
GoogleCodeExporter
closed
9 years ago
9
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