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#163 No audio (in and out)

Obsolete
nobody
None
Medium
Defect
2011-06-17
2010-08-24
Anonymous
No

Originally created by: yaleks...@gmail.com
Originally owned by: r3gis...@gmail.com

What steps will reproduce the problem?
1.Make a call
2.Recipient pick up the phone
3.There is no audio in/out

What is the expected output? What do you see instead?
Expected audio between both parties.
Got no audio and no dial tone.

What version of the product are you using? On what operating system?
Motorola Droid, 2.2, sip provider: voipbuster

Please provide any additional information below.
To get audio I have to press and release soft HOLD button at the upper left corner of the app. What interesting is that it happened only when i make domestic call... Everything is working fine when it is international call.

Related

Tickets: #356
Tickets: #760

Discussion

1 2 > >> (Page 1 of 2)
  • Anonymous

    Anonymous - 2010-08-25

    Originally posted by: r3gis...@gmail.com

    Really interesting.

    I'll need a little bit more infos :
    1 - Is bluetooth enabled? (I mean not is there a BT handset, just if BT is activated in android)

    2 - Is that possible for you to provide me some logs :
        * Go to option > settings > Ui> log level (last entry) and set it to 4
        * Reproduce a failing use case
        * Use a logcat app (such as alogcat available on the market) to send me your logs.

    3 - another thing you can try is setting the sip proxy field (see instructions on issue 164 / comment 3 /problem 4

    Owner: r3gis.3R
    Status: Accepted

     

    Related

    Tickets: #164

  • Anonymous

    Anonymous - 2010-08-25

    Originally posted by: alessand...@gmail.com

    I noticed if you don't use international extension even for domestic calls sip wont work.

     
  • Anonymous

    Anonymous - 2010-08-25

    Originally posted by: yaleks...@gmail.com

    Hi,
    I've sent you my phone's log by the aLogcat.

    1. BT was disabled.
    2. Set just like you said.
    3. Tried that with no success.

    Thank you

     
  • Anonymous

    Anonymous - 2010-08-25

    Originally posted by: r3gis...@gmail.com

    @yaleksbox : didn't receive your logs. My email address is r3gis.3r at gmail.com

     
  • Anonymous

    Anonymous - 2010-08-27

    Originally posted by: r3gis...@gmail.com

    According to the logs there is something that finish the call before it is completely established.

    On the log you sent to me, is the in call screen reach the green state (confirmed call / talking)? Or did it directly go from gray (ringing) to red (hangup)?

    Seems the remote endpoint either hangup or decline your call (can be also the sip server that act like that).
    * If you decline the call for the log, please try to take the call and provide me the log with an established communication (a few seconds will be enough).
    * If you always reproduce this use case, the problem is not that there is no audio, but that the call is never established. As it seems to well negociate the codec (it use ilbc - not the best choice to start but should be ok even if choppy - you can try to disable it in the settings to start with a better and reliable codec). I think more that it's your server that doesn't understand the contact you are trying to call. Are you sure your sip server understand the "+" char? some configuration need it to be replaced by 00.

     
  • Anonymous

    Anonymous - 2010-08-27

    Originally posted by: dc3de...@gmail.com

    @yaleksbox make sure you can dial with some otyher phone using the string you are using. I don't know about voipbuster but my provider Callcentric cannot understand the '+'. You must dial domestic (zone 1) calls 1aaannnnnnn and international calls (other zones) 011ccnnnnnnn...

     
  • Anonymous

    Anonymous - 2010-09-02

    Originally posted by: tdbj...@gmail.com

    I've noticed bluetooth has not worked since there were some changes to bluetooth with versions 24 and later, so I am using 23.  I have to switch bt on and off 2 or 3 times but it does work with v12-23. 

     
  • Anonymous

    Anonymous - 2010-10-16

    Originally posted by: gleveill...@gmail.com

    I have the same problem on HTC Desire 2.2
    It rings but when recipient picks up, no audio In or Out.
    If I press the Pause button (pause and unpause), then it works well.
    A bit of echo tho...

     
  • Anonymous

    Anonymous - 2010-10-16

    Originally posted by: r3gis...@gmail.com

    Could you try to activate stun (settings > media > activate stun).

     
  • Anonymous

    Anonymous - 2010-10-17

    Originally posted by: r3gis...@gmail.com

    Any news with latest version (0.00-15 available on the market)?

    Summary: No audio (in and out)

     
  • Anonymous

    Anonymous - 2010-11-01

    Originally posted by: ser...@gmail.com

    I have similar problem (Samsung Galaxy i5800). No sound on call. Also, when a call is in progress, the picture meaning silent mode is seen in upper side of the screen.

     
  • Anonymous

    Anonymous - 2010-11-01

    Originally posted by: ser...@gmail.com

    Sorry, forgot to give more info. Android v.2.1, cSipSimple all versions available for downloading, SIP provider Callcentric.

     
  • Anonymous

    Anonymous - 2010-12-04

    Originally posted by: aamir...@gmail.com

    HEllo, I earlier posted a problem regarding no audio when making
    outgoing calls, now I have also trod some suggestions available here,

    Soft pressing the hold button and them resuming gives sound, other writer
    There is no sound.
    Galaxy s, froyo ,csipsimple,15-17  trying on sipgate
    Bluetooth disabled / switched off from phone.

     
  • Anonymous

    Anonymous - 2011-01-14

    Originally posted by: dku...@gmail.com

    Tried with Motorolla Mailstone (2.1-upgrade-1) & HTC Desire A18181 (2.2)

    on both cases I do not hear anything during the call. Media streams are correct. On log I see error message "AudioMgr Error:Invalid output format flag; disabling PostProcessing"

    I have recorded call - file is correct sound is present.

    Samsyng Galaxy I9000 works fine.

    I am ready to send any logs and make any examples

     
  • Anonymous

    Anonymous - 2011-01-14

    Originally posted by: r3gis...@gmail.com

    @dku : your devices (desire and milstone) are probably affected by the PSP problem.

    For HTC desire, it's highly possible cause it's already known that HTC has PSP behavior when screen goes off.
    In latest dev version (http://nightlies.csipsimple.com/trunk/) it should be correctly auto-detected now and automatically activate the workaround against PSP problem.

    On Milestone I'm less sure. It's maybe PSP. You can try to activate the PSP workaround manually.
    Activate ExpertSettingMode (wiki page => for global settings), and in User interface > activate Keep awake while in call.

    But could also be some routing issue. If so maybe worth to try what is listed here :
    https://code.google.com/p/csipsimple/wiki/FAQ#Audio_routing_troubleshooting
    Audio routing troubleshooting section

    Let me know how it goes.

     
  • Anonymous

    Anonymous - 2011-01-18

    Originally posted by: dku...@gmail.com

    1. I have made tests between Sony Ericsson Xperia X8 & x10
    There are better then on previous tests (HTC & mailstone ) but still not good enough. Will try to make additional tests later.
    2. On my point of view there is not linked with screen off.
    3. Tested under Samsung I550 - works fine.
    4. For HTC & Mailstone tried to play with setting from trouble shouting list.
    No changes.

    How do you think Can it be linked with wrong sound device driver? Possible problem should be sorted out on PJSIP sound device level?

     
  • Anonymous

    Anonymous - 2011-02-20

    Originally posted by: enciso.d...@gmail.com

    I have an HTC Desire running froyo, Csipsimple works only with PBXES.org 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 - 2011-05-16

    Originally posted by: nate.ka...@gmail.com

    I use csipsimple with sipgate.  The default STUN server for sipgate is used and activated, and so is ICE; however, I lose my incoming audio most of the time, although I do briefly get it back here and there. The loss is probably due to changing between wifi and GSM.

    I'd like to try changing the STUN server to see if that helps me out at all. Would any other STUN server work with sipgate?

    (I've got an LG Optimus V.)

    Would it be worth just considering a different voip provider?
    I think I could take any that provide free inbound calls,
    is that the standard - the great majority of them?

     
  • Anonymous

    Anonymous - 2011-05-25

    Originally posted by: teho...@gmail.com

    I use CSIPSIMPLe on Cyanogenmod 7.0.3 with my HTC Desire connecting to my own Asterisk server and I have the exact same issue: although I can register w/o much probblems, I have no incoming sound at all.
    I tried playing with the codecs (I usually prefer using ulaw) but it didn't change anything.
    Then I Also tried to set up stun.3cx.com as a stun server but it didn't resolve the issue.

     
  • Anonymous

    Anonymous - 2011-05-25

    Originally posted by: r3gis...@gmail.com

    @teho : you could maybe try to follow these instructions about routing :
    https://code.google.com/p/csipsimple/wiki/FAQ?wl=en#Audio_routing_troubleshooting

    I'm not sure it will help, but since CM7 on HTC desire audio driver may not integrate the new API for sip calls, you should try to revert to default modes (instead of the one I set when I detect Gingerbread cause I assume all manufacturer did things to support the new audio modes) :
    Micro source : select default instead of communication
    Mode for sip calls : select normal instead of in_communication.

     
  • Anonymous

    Anonymous - 2011-05-25

    Originally posted by: teho...@gmail.com

    Tried changing all those options, tried all of the other choices in both menus, tried changing the API Modes, the Galaxy hack and everything else I could think of (codec rates etc.), nothing would do...
    I also reverted to using a local authentication on the same network as the SIP server (avoiding STUN problems), it didn't change anything.
    According to my asterisk server logs, everything looks fine, sounds are played correctly and there's no visible handshake problem.

    That's really a pity, that softphone really looks incredible, GPL code, loads of options, recording and all I could ever dream of... Damn.
    CM 7.0.3 is Android 2.3.3.

     
  • Anonymous

    Anonymous - 2011-06-07

    Originally posted by: ned...@gmail.com

    I am having the same issue with a Motorola Milestone. Some outgoing numbers work fine but some have no audio in either direction. It seems to be a problem with the integration with the Android Dialer as it does not seem to happen when making calls directly from CSipSimple. I have one number that fails everytime from the dialer, which happens to be a Google Voice number. It does work if I make the call through CSipSimple. I am using Android 2.2.1 and FreePhoneLine.

     
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