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#296 Freephoneline.ca support

Fixed
nobody
None
Low
Defect
2014-02-06
2010-10-22
Anonymous
No

Originally created by: r3gis... (code.google.com)@gmail.com
Originally owned by: r3gis... (code.google.com)@gmail.com

Freephoneline.ca sip provider.

It registers well.

But when you try to make a call there is no audio in both ways.

I need some feedback from users with this provider.
If you get it working could you share your configuration (STUN, ICE, TURN? DNS SRV?)

Thx in advance.

Related

Tickets: #1237
Tickets: #209

Discussion

1 2 > >> (Page 1 of 2)
  • Anonymous

    Anonymous - 2010-11-02

    Originally posted by: samuel.k... (code.google.com)@gmail.com

    When I call out I have sound both ways.
    When I receive calls, I can't hear the other side.  They can hear me.

     
  • Anonymous

    Anonymous - 2010-11-02

    Originally posted by: r3gis... (code.google.com)@gmail.com

    @samuel : STUN is activated?

     
  • Anonymous

    Anonymous - 2010-11-02

    Originally posted by: samuel.k... (code.google.com)@gmail.com

    It works after it's activated.
    Thanks

    How do I keep the connection alive?
    I woke up and test and it can't receive calls

     
  • Anonymous

    Anonymous - 2010-11-02

    Originally posted by: r3gis... (code.google.com)@gmail.com

    @samuel : great news :) Well at least one person for who it's working with freephoneline !

    There is experimental settings that could help to ensure you'll always receive calls.

    (Assuming you have set easy configuration to "Always available" profile)

    You can tweak the account re-registration time and keep alive.
    To do so, while editing the account press "Menu" and then choose wizard. Choose Expert wizard. Then you'll automatically return to the account list and notice that the icon of your account has changed. It's now an expert account (pre-filled with values from the previous wizard). You can edit it. And you'll see now a lot of new fields.

    Some users living in countries with special data coverage reported me that setting these values improve their availability :

    Re-Register (register timeout) : 184
    Keep Alive : 100

    Let me know how it goes, I'd be interested by feedback with these values to add a "High availability profile" in settings that could make easier this process :).

     
  • Anonymous

    Anonymous - 2010-11-02

    Originally posted by: samuel.k... (code.google.com)@gmail.com

    Just did some testing.  I test a couple times.
    I have now enable "always available" but haven't gone into Expert Wizard yet.
    After enabling the STUN SERVER

    On Wifi, everything works fine, like my first reply.
    On 3G/HSPA, the same problem.  On receive calls, I can't hear the other person.

    Let me know if you want me to test anything.

     
  • Anonymous

    Anonymous - 2010-11-03

    Originally posted by: r3gis... (code.google.com)@gmail.com

    Oh ok sad, just as for some other users so.
    In fact, I've opened this issue cause for some freephoneline.ca users there had this issue. So wanted to open a place to share users experience here. Cause for some users it worked and for others not...

     
  • Anonymous

    Anonymous - 2010-11-03

    Originally posted by: lamdog... (code.google.com)@gmail.com

    Just to let you know.

    3CXPhone works on 3G for me. 
    But their program is not refined, yours is much better.

    Update: on 3G
    I tried the 3CXPhone Stun server stun3.3cx.com on CsipSimple and still doesn't work.

     
  • Anonymous

    Anonymous - 2010-12-05

    Originally posted by: paul_...@hotmail.com

    I tried CSIPSimple on my Samsung Galaxy S Captivate and having the same problems with calls through freephoneline.ca over 3G - no audio.  I tried playing around with some of the settings and the only setting that seems to make a difference is the UDP port. 

    I had this same issue using freephoneline service at home through a softphone.  The way it was resolved with the help of freephoneline.ca tech support, was that I had to open UDP ports 5060, 5061, 6060, 6061, 13000, and 13001 on my router.  If not all of these ports were open I only got one-way audio and it seemed to change depending on which port I opened.

    Since you can only specify one UDP port in the settings of CSIPSimple, I wander if this may be causing the issue.  Is there a way to somehow be able to specify 6 UDP ports in the settings?

    Also, could someone may be make a pre-configured account setting for freephoneline.ca in the list of accounts? The sip settings for freephoneline.ca are actually very simple:

    Username: your freephoneline account phone number (1xxxxxxxxxx)
    Password: password you get with your sip settings file
    proxy: voip.freephoneline.ca
    UDP Ports: 5060, 5061, 6060, 6061, 13000, and 13001

    That's all that should be needed for it to work.

    Love the app, too bad it does not work.

    PS. Works fine over wifi becuase home router has proper ports open.
    Not a carrier issue - I'm on Rogers 3G and freephoneline works fine through fring.

     
  • Anonymous

    Anonymous - 2011-01-11

    Originally posted by: pshp... (code.google.com)@gmail.com

    the audio issue appears to have a few different behavior depending on the connectivity option and the phone provider/technology on each end:

    - CSIPSimple over 3G to POTS, CSIPSimple side will not able to direct audio, POTS side can hear the audio
    - CSIPSimple over 3G to 3G on iphone, CSIPSimple side will not able to direct audio, 3G side can hear the audio
    - CSIPSimple over 3G to freephoneline voip on ADSL, audio work bi-directionally

    tested with almost all option combination and still could not have it resolved. linphone appears the only working sipclient on android over freephoneline per our testing, sipdroid, 3cx and some others either have registration issue or audio issue.

     
  • Anonymous

    Anonymous - 2011-01-11

    Originally posted by: r3gis... (code.google.com)@gmail.com

    @pshpang :
    Did you try ICE + STUN on latest dev version (http://nightlies.csipsimple.com/trunk/) ?
    Really sounds a network issue (NAT network cause 3G is natted and need your to activate at least STUN and maybe ICE too). Cause if it works on wifi it mean that the problem doesn't come from the device itself but to the way CSipSimple is announcing it's own IP address.
    In a NATed network, it require the help of a working STUN server to announce a correct IP. In others android sip client stun is by default activated but I do not cause else problem is worse for people in russia, in china and in other countries where the public network is not directly plugged to the public network where the default STUN server is.

     
  • Anonymous

    Anonymous - 2011-02-20

    Originally posted by: enciso.d... (code.google.com)@gmail.com

    I have an HTC Desire running froyo, Csipsimple works only with PBXES.org(google voice trunk) as voip provider, my regular provider is freephoneline.ca I couldn't make work as always not sound in or outbound, nimbuzz on the other hand works well very good sound, the only problem is that gets disconnected from wifi after a while, so inbound calls don't get through. Help Please, I want this to work!

     
  • Anonymous

    Anonymous - 2011-02-20

    Originally posted by: r3gis... (code.google.com)@gmail.com

    @enciso : useless to post everywhere :) This issue is the right place :).
    I guess from the begin that it is something about some missing setting. If somebody find something that works. (I guess something with stun or stun+ice), I'll create a wizard for that.

    As for nimbuzz, the fact it works is absolutely not relevant : nimbuzz use a proprietary protocol between you and their server ! And then from nimbuzz server it ask the freephoneline.ca server for the call. As consequence it works just like if you configure pbxes.org to have a trunk with freephoneline.ca. So no direct connection and as consequence the fact it works is not relevant.

    Usually in this case it's cause of the fact in 3G you are in a NATed network. As consequence as written in the FAQ, it could be useful to activate STUN. I don't know if freephoneline.ca has its own stun server but if so it could help to use their stun server.

    @pshp... : You can also try to change rtp port. By default it is 4000 but these port are maybe blocked by your 3G provider. (It's in ExpertModeSettings in Network > RTP port).

     
  • Anonymous

    Anonymous - 2011-03-11

    Originally posted by: leper1... (code.google.com)@gmail.com

    Anyone get this resolved? I dont receive any audio via telus 3g :(

     
  • Anonymous

    Anonymous - 2011-03-21

    Originally posted by: mahir... (code.google.com)@gmail.com

    Hello Everybody,

    Last night I installed new codecs as freephoneline preferred codec is G729a which is not installed by default. After doing so and after enabling STUN and ICE I am able to talk through my Home Cable Internet.

    I have already opened the ports 13000,13001,5060,5061,6060,6061 in my home router which is DIR-655.

    But I am still unable hear the second party while using the BELL 3G network.

    If you guys know anything about that, let me know.

    Hope my investigation helps.

    Regards
    Mahir

     
  • Anonymous

    Anonymous - 2011-03-21

    Originally posted by: mahir... (code.google.com)@gmail.com

    Hello Guys

    Problem solved

    I called tech supp. of freephone, they change the user agent to CSIPSimple and it worked.

    Thanks for everybody support.

    Regards
    Mahir

     
  • Anonymous

    Anonymous - 2011-03-22

    Originally posted by: r3gis... (code.google.com)@gmail.com

    Thx a lot Mahir for sharing the info !

    That's an excellent news :).

    I'm not sure to understand what they changed on their side to make it working but... if it works it's the most important :).

    However as I'm not sure what they did on their side, it could be interesting if they can contact me to say what was wrong. As I'll possible change the CSipSimple user-agent to add more info (such as android version and device -- that's requested by some users). I'd prefer not to break they fix freephonline.ca did on their side.
    However, as I'll not do this change soon, for now it's fine ;). (And anyway I think that I'll add an option to customize useragent when I'll implement it so that it will be easy to workaround)

    Status: Fixed

     
  • Anonymous

    Anonymous - 2011-05-19

    Originally posted by: derauco... (code.google.com)@gmail.com

    Can you please give me the exact settings you have for freephoneline.ca using cSIPSIMPLE, I want to set this up but up until now I have had no sucess.
    my email is jackofallsortss@gmail.com  Thanks

     
  • Anonymous

    Anonymous - 2011-07-25

    Originally posted by: qizho... (code.google.com)@gmail.com

    Can you please give me the exact settings you have for freephoneline.ca using cSIPSIMPLE, my email is qizhongy@gmail.com  Thanks

     
  • Anonymous

    Anonymous - 2011-07-26

    Originally posted by: shkai... (code.google.com)@gmail.com

    Can you please send me the cSIPSIMPLE settings for freephoneline.ca i did setup every thing but other party cannot hear me and i can hear the other party.

    my email is aijaz@freshideas4u.com
    Thanks

     
  • Anonymous

    Anonymous - 2011-08-30

    Originally posted by: r3gis... (code.google.com)@gmail.com

    Thanks to new feedback from Ted, (and his very valuable wireshark trace) I think that now know why you are getting that and why other get rid of that.

    To make it quick the way to solve is to disable all codecs but one.
    Go in Settings > Media > Codec
    And long press on all codecs but only one (for example g729) and disable these codecs.
    I would advise to activate only G711 for wideband tab and only g729 for narrowband tab.
    Or you can leave activated only g729 for both tabs if you really want to use g729 even on wifi.

    The important point is to leave *only* one codec activated.

    Copy of the reply I made to Ted to explain why you are getting that and how to solve.
    ----
    I think that I got it, there is a missing feature on freephoneline.ca that produce this apparent bug on csipsimple.
    I'll try to explain what is happening :
    First point to know is that pjsip (the sip stack on which csipsimple rely), only support one codec at a time when the call is established. The problem is that freephoneline.ca reply with several codecs (g711 and g729). It is absolutely valid regarding SIP RFC, and pjsip at this point is ok with that. The point, is that then, in order to fix only one codec for this call, pjsip send a re-invite/update in order to have only one codec (the one prefered in your list). So it send this re-invite/update and if freephoneline.ca server where working properly they should then only use one codec, the one selected by pjsip !
    Instead it is buggy ! It still send with the other codec... So bug on the server side.
    I asked to pjsip guys to try to support multiple codecs at a time which would avoid this reinvite and solve issue with this buggy servers. But sounds not so easy on their side. So, for now, the only solution is to disable all codec but the one you prefer. By the way on this server, it's not a big problem cause you know exactly what are supported codecs. So, you should try to disable all codecs but g729 in codec settings of csipsimple and you'll probably get two way audio.

     
  • Anonymous

    Anonymous - 2011-08-30

    Originally posted by: r3gis... (code.google.com)@gmail.com

    Based on Ted's feedback, I've just created in [r1017] a freephoneline.ca wizard.
    (sounds it's also necessary to disable the contact rewrite feature of pjsip).

     

    Related

    Commit: [r1017]

  • Anonymous

    Anonymous - 2011-09-25

    Originally posted by: beaulieu... (code.google.com)@gmail.com

    SIP on my android phone (Samsung Galaxy S)
    I have tried many applications, free and not, and the only one Ici found  with sound working both ways with freephoneline.ca is Nimbuzz.

     
  • Anonymous

    Anonymous - 2012-02-07

    Originally posted by: jonathan... (code.google.com)@gmail.com

    I can confirm nimbuzz works when csipsimple does not.

     
  • Anonymous

    Anonymous - 2012-04-02

    Originally posted by: CPU.Gast... (code.google.com)@gmail.com

    Hello, I try to use the G729 codec, but it doesn't work, I always have to use PCMU instead and it is not the best to use over 3G.  Me too I'm on Freephoneline.  The only codec I enabled is G729 as you said, but when I call, it failed, when I call from my home phone, it goes directly to voicemail.

    Do I need to open a new ticket ?

     
  • Anonymous

    Anonymous - 2012-11-28

    Originally posted by: callcent... (code.google.com)@gmail.com

    I installed csipsimple on my android samsung y 5360 for using freephoneline.ca. I have already account of freephoneline which i am using on my laptop. On my samsung device it shows registering since a week but its not registered. Please resolve my problem.i also installed hotspot shield on my device to activte it but still it shows registering. Please Send me settings on my email:callcentre786@gmail.com

     
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