No sound after first call on ARM embedded platform

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Hi Anne,

I had issues with ALSA a while back, mainly sample rates required by the CODEC selected for a given call.
After I disabled portaudio altogether the CODEC sampling rate problems went away.

To disable portaudio (at least how I did it) I renamed the /etc/asound.conf file to something like
/etc/asound.conf_DISABLE and rebooted the device.

Might be worth a try, please let me know if that works ( I save information like this :))

Regards,
Martin Woscek
mwoscek at mitre.org<mailto:mwoscek at mitre.org>
Office:703-983-2650
FAX:   703-983-7142

From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of Anne Potgieter
Sent: Thursday, April 11, 2013 2:45 AM
To: pjsip at lists.pjsip.org
Subject: No sound after first call on ARM embedded platform

Hi,

We are trying to embed PJSIP within our embedded application on a Variscite AM3517 ARM based Linux V2.6.34 board. We have done this on a different platform before using PJSIP 1.x without any problems. Unfortunately we have a problem with the latest PJSIP V2.1. We can only establish sound using our first call. When we disconnect this call and redial (from our phone to the PJSIP ARM device) the SIP call is established correctly (we can send and receive DTMF and there are no errors) but there is no sound from our speaker and microphone. We have tested the same application on a X86 Linux machine without any problems. It looks like PJSIP is not opening the ALSA device correctly for the second time. We can establish a single call again after a full restart of the application.

We are using ALSA with the PortAudio configuration. Our microphone is connected to the line-in of the board. The sound tests of PJSIP returns without any problems (we can even play sound twice). Playing/recording sound using ALSA (aplay, arecord etc.) as not revealed any problems. I have included a log with some PJSIP debug information.

Could anyone tell us what the problem could be?

Thanks,

Anne
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