Home / server / Questions / 810257
Accepted
markhorrocks
Asked: 2016-10-21 06:38:14 +0800 CST

Configure asterisk for blink and linphone

  • 772

I have setup an Asterisk server (14.0.2) on Ubuntu 14.04. I can get sound from Twilio using ulaw and from Zoiper (no STUN or ICE). In each case the asterisk server plays gsm files.

I can't get any sound from either Linphone or Blink software phones although both register fine. These are installed on an Ubuntu 16.04 laptop (Dell Inspiron-13-7359). I have enabled every codec available on the phones and tried all available audio device settings for my laptop. Any help would be appreciated.

Here is my Asterisk PJSIP config.

[transport-udp-nat]
type = transport
protocol = udp
bind = 0.0.0.0
; NAT settings
local_net = 10.0.0.0/8
external_media_address = 12.345.67.254
external_signaling_address = 12.345.67.254
[endpoint-internal](!)
type = endpoint
transport = transport-udp-nat
context = Local
disallow=all
allow=ulaw
allow=alaw
allow=slin
allow=g729
allow=g722
allow=opus
allow=gsm
direct_media = no
trust_id_outbound = yes
device_state_busy_at = 1
dtmf_mode = rfc4733

Here is core show codecs in asterisk (redacted)

       3 audio alaw         alaw             (G.711 a-law)
      19 audio speex        speex            (SpeeX)
      20 audio speex        speex16          (SpeeX 16khz)
      21 audio speex        speex32          (SpeeX 32khz)
      23 audio g722         g722             (G722)
      18 audio g729         g729             (G.729A)
       8 audio slin         slin             (16 bit Signed Linear PCM)
       9 audio slin         slin12           (16 bit Signed Linear PCM (12kHz))
      10 audio slin         slin16           (16 bit Signed Linear PCM (16kHz))
      11 audio slin         slin24           (16 bit Signed Linear PCM (24kHz))
      12 audio slin         slin32           (16 bit Signed Linear PCM (32kHz))
      13 audio slin         slin44           (16 bit Signed Linear PCM (44kHz))
      14 audio slin         slin48           (16 bit Signed Linear PCM (48kHz))
      15 audio slin         slin96           (16 bit Signed Linear PCM (96kHz))
      16 audio slin         slin192          (16 bit Signed Linear PCM (192kHz))
       2 audio ulaw         ulaw             (G.711 u-law)
       4 audio gsm          gsm              (GSM)
redundancy)
  28 audio opus         opus             (Opus Codec)
ubuntu asterisk sip codecs

1 Answers

  1. Best Answer

    needed rtp_symmetrical = yes in the endpoint config.

    also didn't work until I removed the gsm codec, possibly too many codecs for one endpoint.

    • 0