C - how to resolve this error while using pjsip?

I am getting this error while using pjsip. How to resolve this? i have mic/speaker in the system but its failing to get the device.

http://paste.ubuntu.com/6504337/

/* Create audio device parameter to open the device */
static pj_status_t create_aud_param(pjmedia_aud_param *param,
                    pjmedia_aud_dev_index capture_dev,
                    pjmedia_aud_dev_index playback_dev,
                    unsigned clock_rate,
                    unsigned channel_count,
                    unsigned samples_per_frame,
                    unsigned bits_per_sample)
{
    pj_status_t status;

    /* Normalize device ID with new convention about default device ID */
    if (playback_dev == PJMEDIA_AUD_DEFAULT_CAPTURE_DEV)
    playback_dev = PJMEDIA_AUD_DEFAULT_PLAYBACK_DEV;

    /* Create default parameters for the device */
    status = pjmedia_aud_dev_default_param(capture_dev, param);
    if (status != PJ_SUCCESS) {
    pjsua_perror(THIS_FILE, "Error retrieving default audio "
                "device parameters", status);
    return status;
    }
    param->dir = PJMEDIA_DIR_CAPTURE_PLAYBACK;
    param->rec_id = capture_dev;
    param->play_id = playback_dev;
    param->clock_rate = clock_rate;
    param->channel_count = channel_count;
    param->samples_per_frame = samples_per_frame;
    param->bits_per_sample = bits_per_sample;

    /* Update the setting with user preference */
#define update_param(cap, field)    \
    if (pjsua_var.aud_param.flags & cap) { \
        param->flags |= cap; \
        param->field = pjsua_var.aud_param.field; \
    }
    update_param( PJMEDIA_AUD_DEV_CAP_INPUT_VOLUME_SETTING, input_vol);
    update_param( PJMEDIA_AUD_DEV_CAP_OUTPUT_VOLUME_SETTING, output_vol);
    update_param( PJMEDIA_AUD_DEV_CAP_INPUT_ROUTE, input_route);
    update_param( PJMEDIA_AUD_DEV_CAP_OUTPUT_ROUTE, output_route);
#undef update_param

    /* Latency settings */
    param->flags |= (PJMEDIA_AUD_DEV_CAP_INPUT_LATENCY |
             PJMEDIA_AUD_DEV_CAP_OUTPUT_LATENCY);
    param->input_latency_ms = pjsua_var.media_cfg.snd_rec_latency;
    param->output_latency_ms = pjsua_var.media_cfg.snd_play_latency;

    /* EC settings */
    if (pjsua_var.media_cfg.ec_tail_len) {
    param->flags |= (PJMEDIA_AUD_DEV_CAP_EC | PJMEDIA_AUD_DEV_CAP_EC_TAIL);
    param->ec_enabled = PJ_TRUE;
    param->ec_tail_ms = pjsua_var.media_cfg.ec_tail_len;
    } else {
    param->flags &= ~(PJMEDIA_AUD_DEV_CAP_EC|PJMEDIA_AUD_DEV_CAP_EC_TAIL);
    }

    /* VAD settings */
    if (pjsua_var.media_cfg.no_vad) {
    param->flags &= ~PJMEDIA_AUD_DEV_CAP_VAD;
    } else {
    param->flags |= PJMEDIA_AUD_DEV_CAP_VAD;
    param->vad_enabled = PJ_TRUE;
    }

    return PJ_SUCCESS;
}

ERROR:

14:13:41.786    pjsua_aud.c  ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
Exception: Object: {Account <sip:192.168.1.16:60791>}, operation=make_call(), error=Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV)

EDIT:

Check if my system has playback and capture device or not (as you can see following shows 100% its working without pjsip):

sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: PCH [HDA Intel PCH], device 0: CA0132 Analog [CA0132 Analog]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0]
  Subdevices: 0/1
  Subdevice #0: subdevice #0
card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ cat /proc/asound/cards
 0 [PCH            ]: HDA-Intel - HDA Intel PCH
                      HDA Intel PCH at 0xd2710000 irq 47
 1 [NVidia         ]: HDA-Intel - HDA NVidia
                      HDA NVidia at 0xd1000000 irq 17
 2 [U0x46d0x825    ]: USB-Audio - USB Device 0x46d:0x825
                      USB Device 0x46d:0x825 at usb-0000:00:14.0-4, high speed


sun@sun-M14xR2:/var/tmp/pjproject-2.1.0/pjsip/src/pjsua-lib$ gst-launch-0.10 -v alsasrc device=hw:2 ! audioresample ! audio/x-raw-int,rate=48000 ! autoaudiosink
Setting pipeline to PAUSED ...
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000
/GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstAudioResample:audioresample0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstCapsFilter:capsfilter0.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse.GstPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)48000, channels=(int)1
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 1.000000
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: volume = 1.000000
/GstPipeline:pipeline0/GstAutoAudioSink:autoaudiosink0/GstPulseSink:autoaudiosink0-actual-sink-pulse: mute = FALSE
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough
Additional debug info:
gstbaseaudiosrc.c(840): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 10560 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
WARNING: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough
Additional debug info:
gstbaseaudiosrc.c(840): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 9600 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.

Your problem is related to audio system. Most Linux systems run PulseAudio over Alsa, like yours (you can see this in GStreamer's logs) but pjsip by default enables PortAudio driver for Linux.

To fix it, you can enable an available Alsa driver by adding:

#define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1

To pjlib/include/pj/config_site.h. If it does not exist, you can create it like:

#define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1
#include <pj/config_site_sample.h>

And rebuild (you can directly rebuild only pjmedia: run make at pjmedia/build folder).

Note: It may be possible that you've to disable currently configured driver by editing pjmedia/build/os-linux.mak and setting AC_PJMEDIA_SND to a different value (e.g. alsa)