gigincg/sipml5

Issues

• call disconnection in api version SIPML5 API version = 1.5.230

  #229 opened 9 years ago by GoogleCodeExporter
  0

• Temporary failure in name resolution

  #226 opened 9 years ago by GoogleCodeExporter
  0

• Call.info() result is logged to console instead of given to user

  #227 opened 9 years ago by GoogleCodeExporter
  1

• One way video, any browser, webrtc2sip and Asterisk to softphone

  #228 opened 9 years ago by GoogleCodeExporter
  2

• Parameters send to Event Listener Function

  #223 opened 9 years ago by GoogleCodeExporter
  2

• Outbound call hangs up after 1 minute with FreeSwitch Session-Expire

  #224 opened 9 years ago by GoogleCodeExporter
  0

• Problem in BYE send from SIPML5 - Webrtc2sip (Version 2.6.0)

  #225 opened 9 years ago by GoogleCodeExporter
  0

• sipml5 unable to play remote audio

  #221 opened 9 years ago by GoogleCodeExporter
  2

• No early audio in outgoing sipml call to an extension.

  #222 opened 9 years ago by GoogleCodeExporter
  2

• Firefox ICE resolution to incorrect 0.0.0.0 address

  #217 opened 9 years ago by GoogleCodeExporter
  2

• Video stream doesnt add in SipML5

  #218 opened 9 years ago by GoogleCodeExporter
  0

• Export to GitHub

  #219 opened 9 years ago by GoogleCodeExporter
  0

• Available on main package managers

  #220 opened 9 years ago by GoogleCodeExporter
  0

• Content length for SIP INFO packets is incorrect in latest release

  #216 opened 9 years ago by GoogleCodeExporter
  1

• startBfcpShare not working

  #215 opened 9 years ago by GoogleCodeExporter
  0

• Called in wrong state: STATE_INPROGRESS

  #214 opened 9 years ago by GoogleCodeExporter
  7

• Sipml5 not connecting

  #211 opened 9 years ago by GoogleCodeExporter
  1

• No Audio on Firefox 34+

  #212 opened 9 years ago by GoogleCodeExporter
  0

• dtmf('#') and dtmf('*') are not detected correctly

  #213 opened 9 years ago by GoogleCodeExporter
  0

• Double call "onGetUserMediaSuccess" and "createAnswer". Sipml5 can't answer on some calls and sends BYE.

  #209 opened 9 years ago by GoogleCodeExporter
  4

• SUBSCRIBE does not contain route headers

  #210 opened 9 years ago by GoogleCodeExporter
  0

• Too large SIP INVITE

  #207 opened 9 years ago by GoogleCodeExporter
  0

• tsip_transac_layer.prototype.cancel_by_dialog incorrectly deletes transactions when there is more than one transaction

  #208 opened 9 years ago by GoogleCodeExporter
  2

• enable_media_stream_cache setting does not work

  #205 opened 9 years ago by GoogleCodeExporter
  2

• dtmf formatted incorrectly

  #206 opened 9 years ago by GoogleCodeExporter
  3

• webrtc4all is not detected

  #203 opened 9 years ago by GoogleCodeExporter
  1

• SIP INFO DTMF not compatible with Asterisk

  #204 opened 9 years ago by GoogleCodeExporter
  1

• Call between chrome to softphone call automatically dropped

  #201 opened 9 years ago by GoogleCodeExporter
  1

• Bad invite send by client on Chrome 39 (linux 64)

  #202 opened 9 years ago by GoogleCodeExporter
  1

• Offer message containing c=IN IP4 0.0.0.0

  #198 opened 9 years ago by GoogleCodeExporter
  3

• Not working in safari 5.1.7

  #199 opened 9 years ago by GoogleCodeExporter
  1

• Compatibility problem with Node-Webkit?

  #200 opened 9 years ago by GoogleCodeExporter
  0

• SipML5 + Asterisk : Remote host can't match request ACK

  #197 opened 9 years ago by GoogleCodeExporter
  4

• WebRTC video call rejected asterisk11.130+sipml5

  #195 opened 9 years ago by GoogleCodeExporter
  1

• Is 407 supported by sipml5

  #196 opened 9 years ago by GoogleCodeExporter
  2

• Race condition in hold/resume

  #193 opened 9 years ago by GoogleCodeExporter
  1

• which server can work with sipML5 api

  #194 opened 9 years ago by GoogleCodeExporter
  0

• SIPml5 behind NAT, Freeswitch on Amazon EC2, Calll Hang Up

  #191 opened 9 years ago by GoogleCodeExporter
  2

• sip call from android uses loud speaker instead of front speaker.

  #192 opened 9 years ago by GoogleCodeExporter
  2

• SUBSCRIBE request fails \failing to find dialog

  #189 opened 9 years ago by GoogleCodeExporter
  7

• Asterisk on AWS; RTP reciving at client end but no voice

  #190 opened 9 years ago by GoogleCodeExporter
  1

• No video on Firefox 30 :onIceGatheringCompleted but no local sdp request is pending

  #188 opened 9 years ago by GoogleCodeExporter
  0

• UPD flows bidirectionally but one way audio on webphone

  #185 opened 9 years ago by GoogleCodeExporter
  3

• Incomming call without Video payload is refused

  #186 opened 9 years ago by GoogleCodeExporter
  0

• Settings ice_servers to [] does not disable ICE

  #187 opened 9 years ago by GoogleCodeExporter
  3

• Unable to Login to OfficeSip Server using chrome v. 32

  #184 opened 9 years ago by GoogleCodeExporter
  1

• Chrome m35(forces DTLS by default) <-> asterisk 11.9 [fix]

  #183 opened 9 years ago by GoogleCodeExporter
  6

• Transfer call functionality is not working as expected in sipml5

  #181 opened 9 years ago by GoogleCodeExporter
  1

• Sipml5 SIP OPTIONS response SIP/2.0 405 Method Not Allowed

  #182 opened 9 years ago by GoogleCodeExporter
  0

• WebRTC Not working with Google Chrome 35.

  #180 opened 9 years ago by GoogleCodeExporter
  9