Transfer call functionality is not working as expected in sipml5

GoogleCodeExporter commented 9 years ago

What steps will reproduce the problem?
1. Initiate a call from A(browser client App) to B(Desktop soft phone)
2. Answer the call from B
3. Transfer the call from B to C(Desktop soft phone)

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

Expected : call should be transferred to C ,and B should be free.

Current output : As I initiate the transfer call functionality, below mentioned 
events displayed in browser console.But no ringing at C and also no logs 
related to this transfer request in asteriskLogs

=>State machine: x0000_Any_2_Any_X_i1xx
=>session event = o_ect_trying
=>session event = i_ao_request
=>State machine: tsip_transac_ist_Accepted_2_Terminated_timerL

What version of the product are you using? On what operating system?
Asterisk :11.7.0
webrtc2sip: 2.6.0
OS: Ubantu 64 bit

Please provide any additional information below.

All other APIs like call, answer, hold, unhold, hangup is working properly but 
not transfer

PFA the html file, SIPml5 js file used for the same

Original issue reported on code.google.com by yogender...@gmail.com on 26 May 2014 at 9:45

Attachments:

GoogleCodeExporter commented 9 years ago

I am also facing this issue. Has anybody got a solution or a work around for 
this as I need some solution on urgent basis.

Original comment by anytimea...@gmail.com on 13 Feb 2015 at 8:30

arturkarczmarczyk commented 8 years ago

Anyone got a solution for this problem yet? We try to do the call transfer, but no luck so far.

madhusudangoudk commented 7 years ago

Please let me know ,did anybody got the solution

bizzr3 commented 7 years ago

same here ! no music on hold and also transfer functionality is not work at all. tried with different JS sip stacks but no results.

jakesays-old / sipml5

Transfer call functionality is not working as expected in sipml5 #181