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)
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.
Attribution
Source : Link , Question Author : markhorrocks , Answer Author : markhorrocks