Benny,
I can now see the Bluetooth headset as a plugin device when I run sndinfo,
and I have attempted to run pjsua with this device. This causes pjsua to
core dump. Is this an issue with pjsua or portaudio? I have pasted the error
below and attached to this email:
13:46:00.109 os_core_unix.c pjlib 0.8.0 for POSIX initialized
13:46:00.110 sip_endpoint.c Creating endpoint instance...
13:46:00.111 pjlib select() I/O Queue created (0x816d67c)
13:46:00.111 sip_endpoint.c Module "mod-msg-print" registered
13:46:00.111 sip_transport. Transport manager created.
13:46:00.111 sip_endpoint.c Module "mod-pjsua-log" registered
13:46:00.111 sip_endpoint.c Module "mod-tsx-layer" registered
13:46:00.111 sip_endpoint.c Module "mod-stateful-util" registered
13:46:00.111 sip_endpoint.c Module "mod-ua" registered
13:46:00.111 sip_endpoint.c Module "mod-100rel" registered
13:46:00.111 sip_endpoint.c Module "mod-pjsua" registered
13:46:00.111 sip_endpoint.c Module "mod-invite" registered
13:46:00.698 pasound.c PortAudio sound library initialized, status=0
13:46:00.698 pasound.c PortAudio host api count=2
13:46:00.698 pasound.c Sound device count=14
13:46:00.699 pjlib select() I/O Queue created (0x81c031c)
13:46:00.699 sip_endpoint.c Module "mod-evsub" registered
13:46:00.699 sip_endpoint.c Module "mod-presence" registered
13:46:00.699 sip_endpoint.c Module "mod-refer" registered
13:46:00.699 sip_endpoint.c Module "mod-pjsua-pres" registered
13:46:00.699 sip_endpoint.c Module "mod-pjsua-im" registered
13:46:00.699 sip_endpoint.c Module "mod-pjsua-options" registered
13:46:00.699 pjsua_core.c 1 SIP worker threads created
13:46:00.699 pjsua_core.c pjsua version 0.8.0 for i686-pc-linux-gnu
initialized
13:46:00.700 pjsua_core.c SIP UDP socket reachable at 172.29.70.120:5061
13:46:00.700 udp0x81db024 SIP UDP transport started, published address is
172.29.70.120:5061
13:46:00.700 pjsua_acc.c Account <sip:172.29.70.120:5061;transport=UDP>
added with id 0
13:46:00.700 tcplis:5061 SIP TCP listener ready for incoming connections
at 172.29.70.120:5061
13:46:00.700 pjsua_acc.c Account <sip:172.29.70.120:5061;transport=TCP>
added with id 1
13:46:00.700 pjsua_media.c RTP socket reachable at 172.29.70.120:4000
13:46:00.700 pjsua_media.c RTCP socket reachable at 172.29.70.120:4001
13:46:00.700 pjsua_media.c RTP socket reachable at 172.29.70.120:4002
13:46:00.700 pjsua_media.c RTCP socket reachable at 172.29.70.120:4003
13:46:00.700 pjsua_media.c RTP socket reachable at 172.29.70.120:4004
13:46:00.700 pjsua_media.c RTCP socket reachable at 172.29.70.120:4005
13:46:00.700 pjsua_media.c RTP socket reachable at 172.29.70.120:4006
13:46:00.700 pjsua_media.c RTCP socket reachable at 172.29.70.120:4007
13:46:00.700 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@16000 Hz
13:46:00.702 pjsua_media.c ..failed: Invalid sample rate
13:46:00.702 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@22050 Hz
13:46:00.703 pjsua_media.c ..failed: Invalid sample rate
13:46:00.703 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@44100 Hz
13:46:00.705 pjsua_media.c ..failed: Invalid sample rate
13:46:00.705 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@48000 Hz
13:46:00.706 pjsua_media.c ..failed: Invalid sample rate
13:46:00.706 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@11025 Hz
13:46:00.708 pjsua_media.c ..failed: Invalid sample rate
13:46:00.708 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@32000 Hz
13:46:00.709 pjsua_media.c ..failed: Invalid sample rate
13:46:00.709 pjsua_media.c pjsua_set_snd_dev(): attempting to open devices
@8000 Hz
13:46:01.761 echo_speex.c Speex Echo canceller/AEC created,
clock_rate=8000, samples per frame=80, tail length=200 ms, latency=119 ms
>>>>
Account list:
[ 0] <sip:172.29.70.120:5061;transport=UDP>: does not register
Online status: Online
*[ 1] <sip:172.29.70.120:5061;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: |
Account: |
| |
| |
| m Make new call | +b Add new buddy .| +a Add new
accnt |
| M Make multiple calls | -b Delete buddy | -a Delete
accnt. |
| a Answer call | i Send IM | !a Modify
accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr
(Re-)register |
| H Hold call | u Unsubscribe presence | ru
Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next
ac.|
| U send UPDATE | T Set online status | < Cycle prev
ac.|
| ],[ Select next/prev call
+--------------------------+-------------------+
| x Xfer call | Media Commands: | Status &
Config: |
| X Xfer with Replaces |
| |
| # Send RFC 2833 DTMF | cl List ports | d Dump
status |
| * Send DTMF with INFO | cc Connect port | dd Dump
detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump
config |
| | V Adjust audio Volume | f Save
config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save
config |
+------------------------------+--------------------------+-------------------+
| q QUIT sleep N: console sleep for N ms n: detect NAT
type |
+=============================================================================+
You have 0 active call
>>>
pjsua-i686-pc-linux-gnu:
src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c:2722:
PaAlsaStream_GetAvailableFrames: Assertion `queryCapture || queryPlayback'
failed.
Aborted (core dumped)
Date: Sun, 13 Apr 2008 23:21:56 -0400
> From: "Benny Prijono" <bennylp@xxxxxxxxx>
> Subject: Re: Extended ALSA API
> To: "pjsip list" <pjsip at lists.pjsip.org>
> Message-ID:
> <1879720d0804132021s7e152316i96bef2304f9f5b1f at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On Sat, Apr 12, 2008 at 6:34 PM, Joshua Bryant <josh at bryantweb.com> wrote:
> > No, PJSIP does not seem to see plugin devices defined in the .asoundrc
> file.
> > The extended alsa api is required to see plugin devices. A bluetooth
> > headset in BlueZ is defined as a plugin device in the ~/.asoundrc file
>
> Erm, sorry I'm totally unfamiliar with how ALSA is supposed to work
> with extended API. We're using PortAudio in PJSIP and just use
> whatever PortAudio gives us (unless if we use PortAudio incorrectly of
> course). Sorry can't help more.
>
> Cheers
> Benny
>
>
>
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