Hi
recently i tried to use baresip to call using my router, here it is :
root@Pulpstone-OpenWrt:~# baresip -f /etc/baresip
baresip v0.5.9 Copyright (C) 2010 - 2018 Alfred E. Heggestad et al.
Local network address: IPv4=wlan0:192.168.3.133
evdev: device name: C-Media Electronics Inc. USB PnP Sound Device
Supported event types:
Event type 0x00 (Unknown event type: 0x0000)
Event type 0x01 (Keys or Buttons)
Event type 0x04 (Something miscellaneous)
Supported Keys:
Key 0x71 (Mute)
Key 0x72 (Volume Down)
Key 0x73 (Volume Up)
Key 0xa3 (Next Song)
Key 0xa4 (Play and Pause)
Key 0xa5 (Previous Song)
Key 0xa6 (Stop CD)
Supported LEDs:
aucodec: PCMU/8000/1
aucodec: PCMA/8000/1
ausrc: alsa
auplay: alsa
medianat: stun
medianat: turn
account: username:password is now deprecated please use ;auth_pass=xxx instead
Populated 1 account
Populated 2 contacts
Populated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
baresip is ready.
201@192.168.3.5: {0/UDP/v4} 200 OK (Asterisk PBX 15.3.0) [1 binding]
All 1 useragent registered successfully! (17 ms)
but at my phone, i cant hear any mic input from the router, is there anything wrong???
here is my config file
poll_method epoll
input_device /dev/input/event0 #eventually adapt input device path for your system
input_port 5555
sip_trans_bsize 128
audio_player alsa,default # audio speaker device
audio_source iec958:CARD=Device # audio microphone device
audio_alert alsa,default # audio ring device
#if you want to use more than one audio device you need to use: alsa,default:CARD=devicename
#to discover device name use command: aplay -L
#-----------------------------------------------------------
audio_srate 8000-48000
audio_channels 1-2
rtp_tos 184
rtcp_enable yes
rtcp_mux no
#jitter_buffer_delay 15-35 # uncomment this line only in case you use an external voip provider and you experience high ping latence
rtp_stats no
dns_server 8.8.8.8:53 # use your preferred DNS server
module_path /usr/lib/baresip/modules
module stdio.so
module evdev.so
module g711.so
module alsa.so
module stun.so
module turn.so
module_tmp account.so
module_app contact.so
module_app menu.so
natbd_server <201:201@192.168.3.5> # I'm not sure this line is necessary when you set "stunserver" option on user account
# and/or for local accounts: <user:password@localhost>
natbd_interval 600 # same as above, you might try to comment "#" both lines and test if baresip works
Im using usb sound as my input
thanks in advanced