> Attention!

New threads need to be created only inroot partition! In the future, they will be processed by moderators.

If you posted a new version of the program, please notify the moderator by clicking the "Complaint" button on your message.

Catalog of Android programs



3CXPhone for Android | VoIP telephony



Rep: (242)
3CXPhone for Android
Version: 3.0.50

Last update of the program in the header:13.01.2018

Attached ImageAttached Image

Short description:
VoIP telephony

Description:
3CXPhone for Android A free VoIP Softphone that works with SIP IP PBXs based on 3CX or Asterisk, and also works with all popular VoIP providers. 3CXPhone is not tied to a specific VoIP service.
The main feature of this VoIP Softphone is that the user can create many profiles with different settings and easily switch between them.

Homepage: http://www.3cx.com/blog/3cxphone/android/
Cyrket: http://www.cyrket.com/p/android/com.tcx.sip.ui/
Android Market: https://market.android.com/details?id=com.tcx.sip.ui
market: // search? q = pname: com.tcx.sip.ui

Russian interface: Not
Download:
Version: 3.0.50 rus 3CXPhone for Android (Post lewonchik # 33945215)
Version: 2.0.4 3CXPhone_2.0.4.apk

Past versions


Post has been editedsimorg13 - 13.01.18, 22:07
Reason for editing: new version



Rep: (586)
voice like normal but how to win an echo from one who is on the line, that is, if I'm calling from the 3CXPhone for Android on GSM hear me well but at that one I'm calling hear yourself when zvanok with 3CXPhone for Android on 3CXPhone windows everything is normal



Rep: (5)
GerekMDRus, very simple. Install Microphone amplifier to the minimum settings, try to reduce the volume of the speaker, it is better to cuddle it to his ear :)))



Rep: (67)
with prog MultiFon befriended by strange
proxy prescribed by sip:
like works but when activated with prog calling over GSM is a program crashes, and then does not want to regitsya.
VERY loud tone dialing and until the subscriber does not take the tube sound through the speaker phone.
If a program is vyklyuchaesh 1 call is made on gsm and through a set with the speakerphone.



Pliz give clear recommendations for setting up or any other suggestions for use or other softiny



Rep: (67)
Prog blocked FS caller th so after its outgoing throws



Rep: (28)
catmat,
Write how to make friends, I tried not to register with MultiFon and that's it



Rep: (67)
User
7922000000

password
***********

internal server
is empty

Exter.Server
multifon.ru

Stun server
stun.movial.fi (this is not necessarily likely)

in advanced

proxy
sip: 193.201.229.35

port
5060

regist.timeuot
xs that need work but mine with all parameters (worth 300) if the figure it is better to write

Queer-alive interval
xs that need (worth 0)

DTMF
set in-band

nat and ice without ticks

in settings
for each individual

Ate someone be able to make more accurate data write.
Wash the program does not automatically reconnect.
If it throws than any that then to bridge the need to through the PC to enter the fuflofon and ginger it. (If very close the that connects without problems), so with FS caller better not combine until they fix it or another prog to put an incoming call (but it can be only on my phone, try)
I decided to problemmu a little differently. because no to a computer on the street I adjusted linphone
she very quickly invigorates fuflofon but it can not for some reason to expect a call.



Rep: (28)
like to work, I have a mother and caller Sens 3cx intercepts a set of him, but does not store call history,
in the log of the caller numbers are, but do not correlate with the contacts



Rep: (15)
maddiver8,
Thank you, my good man! More than a month no one could answer. I have everything working. The only thing different is the DTMF (RFC stood something there ...). STUN server is not specified. And do not be reconnected because the keep alive interval you 0? Logically, this should be the time in which it checks the connection.



Rep: (67)
We can then try linphone in another topic together to set up?
tried another soft with photos to fill the screen, too, throws, followed must be mentioned vzbadrivaniem



Rep: (28)
linphone I tried with MEGA, he quickly earned all. but I did not check for rekonekta
and I did not like that the room need to gain out of it instead of the standard caller



Rep: (67)
People put the stuffing setings - integration -



Rep: (0)
All the good time of day!
3CX has put a program on Wildfire, everything works fine! Without any problems!
I describe:
In the profile settings (I tried only one profile), hammered:
User .... (that the provider will give)
Password .... (that the provider will give))
Internal and External server .... the host name where you want to connect to register
Stun Server ... (maybe not needed at all, it all depends on the provider) Default stun3.3cx.com
Advanced ... (that is the default) Port-5060 !!! (Traditional port for registration SIP), Reg.timeout 3600 (re-registration period), can be limited to 600, the smaller the better,
Keep-Alive 60 (packet storage interval), DTMF method-RFC-2833 (recognition tone parcels), NAT and ICE are not included
Integration ..... Lock wi-fi NEVER rest on the discretion
Audio settings ... Enable echo cancellation and silence detection (especially affect the quality of communication)
Audio codecs ... include all codecs. In terms of quality codecs can be placed as follows (in order of decreasing quality): G711, gsm ,. According to the used bandwidth (in order:) GSM, G711. About speex I do not know. The A-G711 used lau.Mne like GSM codec.
My conditions: wi-fi network, the use of local SIP station (not the provider) .Problem connectivity should not PCtel to be, and to other providers, too. uiscom only uses port 5060 for registration does not, and 9060 !!! So be careful!
Reconnect stable! Those. going outside the network registration is lost when returning to its network registration is restored automatically (without asking me any questions).
Inbound and outbound communication is working, audibility in both directions is excellent!
When using a GSM channel and an incoming call on SIP, distributed signal on an incoming call to the handset. At the other end heard rebound! On the contrary! It is understandable, because the microphone and loudspeaker only one)))
I think it can be done and the transfer of calls and make conference, in the presence of a second SIP line to your account!
By the way, all the talk, beeps, etc. through the built-in speaker to the ear! And not through the speakerphone!
The program works as it should!

Post has been editedkostik797 - 25.11.10, 13:25



Rep: (9)
all is well, that's just to even turn off the screen, and the big cheeks.



Rep: (0)
At the moment, it is the only one of the tried me SIP clients running on Samsung i5800 (Galaxy 3). Connect it to CallCentric and quietly happy. I understand that the program is now under heavy development, its previous version with my unit did not want to work, as I wrote in the developer forum. After two or three weeks, I was offered to test the beta version with bug fixes my problem. After another week, this bug fix was included in the new release. So that the ratio of developers to their product pleases.
At the moment I have only one question for 3CXPhone - his tie to 12Voip. I tried with the value indicated on thehttp://www.12voip.com/en/sipp.htmlBut without much success. However, I am in these matters until a beginner.



Rep: (13)
I do not know who enjoys what ... None of the Sip-clients did not accept. Explain why 3G, 3CXPhone signal loss after losing touch with MultiFon forever? At least until you come to it with the help of a native client or through BB Sipdroid? If this is the MultiFon, why Sipdroid safely connects all the time? As a result, we have to use 3 customers on different occasions (at Sipdroid their troubles). Believe me, och is not convenient ...

Post has been editedewg1 - 25.01.11, 15:06



Rep: (256)
Tried probably all dialers, focused onCSipSimple . Completely satisfied. Codecs, the API phone profiles - all the beam.



Rep: (177)
Put the optimal settings Sipnet and PCTEL for this bend.
How to overcome that when choosing a room, he substitutes the prefix 00 and therefore can not collect it?



Rep: (0)
Please help me set up SIPNET! CSIPSimple able to adjust. 3CXPhone - does not work yet. Thanks!



Rep: (4)
itinerant_hobo @ 25.03.2011, 21:19*
3CXPhone - it does not work yet

Soherewritten. However, in the program (with the Market) by default for some reason entered as the STUN server-3CXPhone'vsky. It should be just clear this option. it is not necessary for Sipneta.

galaktiker @ 15.10.2010, 11:30*
Install Microphone amplifier to the minimum settings, try to reduce the volume of the speaker, it is better to cuddle it to his ear

I was not able to win in such a way echo "on the other side." Program until demolished.

Post has been editedSahar_spb - 11.04.11, 15:32



Rep: (34)
people who help him to sip account or mail agenta qip adjust.


Full version    

Help     rules

Time is now: 03/05/20, 15:51