naresh924 / csipsimple

Automatically exported from code.google.com/p/csipsimple
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Transfer calls & conference calls #305

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
Please add the ability to transfer and place calls on hold. Thanks.

Original issue reported on code.google.com by j...@iostechs.com on 25 Oct 2010 at 9:35

GoogleCodeExporter commented 9 years ago
Call on hold => Already available : there is a pause button on the left of the 
contact picture.

Transfer & conference : are planned... but as I develop the soft on my own time 
(night and holidays...) it may take a little bit time unless contributors would 
like to that ;)... 
Just as an information, the native sip stack already support these features, 
that ~just~ user interface to add. For conference that's pretty complicated. 
For transfer should not be too complicated... but still work to do.

Original comment by r3gis...@gmail.com on 25 Oct 2010 at 10:55

GoogleCodeExporter commented 9 years ago
Issue 168 has been merged into this issue.

Original comment by r3gis...@gmail.com on 25 Oct 2010 at 11:12

GoogleCodeExporter commented 9 years ago
Issue 308 has been merged into this issue.

Original comment by r3gis...@gmail.com on 26 Oct 2010 at 5:45

GoogleCodeExporter commented 9 years ago
Hello,

We like your product annd its the only one that work for us, but I will like
to know when the call transfer will be available to use?

Original comment by lachance...@gmail.com on 29 Oct 2010 at 12:51

GoogleCodeExporter commented 9 years ago
@denis : I code this app on my free time (nights and holidays) and only for the 
pleasure. I try to do my best to make happy all users, but can't give you any 
roadmap.
This feature has now a high priority but I can't give you any target date. If 
you are hurry... code it yourself ;) you'll be welcome ;).

Original comment by r3gis...@gmail.com on 29 Oct 2010 at 2:45

GoogleCodeExporter commented 9 years ago
Just for information, task has been started.

Transfer (the very first step of transfer UI) is now available in 0.00-15-10. 

To use the first unintuitive UI (don't be afraid, that's absolutely incomplete):
While in call, press menu and choose transfer. A popup appear (clicking on 
search in contact does nothing for now). And there is a debug field that allow 
you to enter a *Complete* sip uri. By complete I mean something like that : 
sip:contact@domain.loc

(anyway, if it's not a valid sip uri, OK will not be enabled ;) ).

Again, it's the very first UI, so useless to say me that's not easy to use ;).

Original comment by r3gis...@gmail.com on 6 Nov 2010 at 6:05

GoogleCodeExporter commented 9 years ago
Issue 391 has been merged into this issue.

Original comment by r3gis...@gmail.com on 19 Nov 2010 at 7:35

GoogleCodeExporter commented 9 years ago
Issue 424 has been merged into this issue.

Original comment by r3gis...@gmail.com on 27 Nov 2010 at 12:18

GoogleCodeExporter commented 9 years ago
Issue 440 has been merged into this issue.

Original comment by r3gis...@gmail.com on 2 Dec 2010 at 6:24

GoogleCodeExporter commented 9 years ago
First of all, thank you for this excellent app... 

Secondly, I am eagerly waiting for the conferencing feature. That will sate a 
chunk of my SIP calling needs. :)

Furthermore, the "transfer call" feature does not seem to work. Can you please 
help me with this? 

 - I am on a call. 
 - Menu, and then Transfer. 
 - I get a pop-up which asks to enter a number. I key in the number and hit OK. 
 - I get the dial pad..???

Am I missing something? When does the call get transferred? (Doesn't seem to 
work even if I key in the number on the dial pad...)

I am using the latest version. (15-18)

Please help. 

Many thanks, once again. Good luck!

Original comment by karts...@gmail.com on 4 Dec 2010 at 10:23

GoogleCodeExporter commented 9 years ago
Issue 466 has been merged into this issue.

Original comment by r3gis...@gmail.com on 8 Dec 2010 at 8:16

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
call transfert with an internal voip server as 3cx is fully functionally, place 
the first call in pause, with keypad add a new call and, when in line with the 
second call, place the icon of the first call on to the second and the call is 
automatically transfert....
with the 3cx server is ok the sms too!!!

thanks for your fantastic software 

Original comment by rcnmecca...@gmail.com on 6 Oct 2011 at 1:06

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
The app works just fine I haven't yet figured out how to transfer the call, if 
I follow the instructions on comment 13 I enter conference call and the call is 
not transfered.
Unless "place the icon of the first call on to the second and the call" is not 
what I am doing...
I have the app installed on my galaxy tab

please help

best to all

Original comment by gomesmac...@gmail.com on 18 Nov 2011 at 3:36

GoogleCodeExporter commented 9 years ago
You need to place the first call on-hold before dialing the second one. Pull 
off that drop down thing under the contact pic and press pause || button. The 
icon will shrink and move to the right corner. Than after dialing the second 
line you will be able to "place the icon of the first call on to the second".

Original comment by hotplug...@gmail.com on 29 Nov 2011 at 6:00

GoogleCodeExporter commented 9 years ago
The app is great. I tried sipdroid and 3cx, but they both had a great 
performance issues on my weak alcatel ot908. On csip everything just rocks. 
It's so awesome that I even decided to do something for you and made a complete 
translation for Poland. There is just one sad thing - I can't transfer calls - 
when I put a 1st call on hold and dial the other one, everything works fine, 
but when I drag the 1st call on second call, csip makes something like 
conference call. I use lots of transfers everyday and lack of this feature 
makes this great app almost unusable in my situation. I can test it for you or 
help in any other way if you like to...

Original comment by staff1...@gmail.com on 19 Dec 2011 at 8:47

GoogleCodeExporter commented 9 years ago
Try to drag the second call on the first call instead of the first on the 
second ;). 
The center of the UI is for current call. The side is for held calls. When you 
make a transfer you transfer the current call to the held call. So drag the one 
from the center to the one on the side of the screen.

Thx for the translation :) I'll try to integrate it as soon as possible.

Original comment by r3gis...@gmail.com on 19 Dec 2011 at 9:08

GoogleCodeExporter commented 9 years ago
I'll try it tomorrow. It sounds completely vice versa to the instructions from 
comment 13 where it is written to put the 1st one on hold and then drag it to 
the 2nd (active) call :)

Original comment by staff1...@gmail.com on 19 Dec 2011 at 9:23

GoogleCodeExporter commented 9 years ago
hi every body :) i'm using your project to develop my own application but i 
have a little problem with the xfer fonction : IT DOES'NT work :/  in my java 
code i wrote this line >> service.xfer(currentCall.getCallId(), callee); where 
the callee is the number to receive the call that i transfered it .. the format 
of the callee is 
sip:601@ server adress

Original comment by mehdi22m...@gmail.com on 20 Jul 2012 at 12:54

GoogleCodeExporter commented 9 years ago
hi medhi, where is hosted your source code so that I can have a look on what 
could go wrong?

Original comment by r3gis...@gmail.com on 20 Jul 2012 at 1:00

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
The transfer works wonderfully well.
The problem is with the conference calls, I am posting a year later from the 
last post here and it seems that it was never really fixed.
I am currently using the latest nightly version of csipsimple (ver 1.01.00 
r2284)

What I am doing:
1) Receive a call.
2) Put it on hold
3) Press the + on the bottom of the screen (Pickup SIP contact) and call a 
second number
4) Resume in both calls.

Both callers can hear my voice, but they can't hear each others.

Original comment by toro...@gmail.com on 9 Sep 2013 at 6:04

GoogleCodeExporter commented 9 years ago
Is there has others way with sip can fix conference calls?

Original comment by tomezho...@gmail.com on 12 Sep 2013 at 7:41

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
About conferencing and can't hear each other, the tracking issue 2027 has just 
been fixed. Next nightly build should work better :)

Original comment by r3gis...@gmail.com on 14 Sep 2013 at 8:47

GoogleCodeExporter commented 9 years ago
I downloaded Head version, "src\org\pjsip" is empty.I hope that there is a hole 
right version which can be used.Very thanks.

Original comment by tomezho...@gmail.com on 16 Sep 2013 at 8:46

GoogleCodeExporter commented 9 years ago
Have a look to the the HowToBuild wiki page.

If you just want to test you can also use the nightly build : 
http://nightlies.csipsimple.com/trunk/

Original comment by r3gis...@gmail.com on 16 Sep 2013 at 10:17

GoogleCodeExporter commented 9 years ago
[deleted comment]
GoogleCodeExporter commented 9 years ago
I tested the transfer method and it doesn't work anyway... I tested with 
CSipSimple-latest-trunk.apk       29-Oct-2013 12:19  6.4M

Original comment by aquilaxxx on 29 Oct 2013 at 10:39