Unable to find a codec translation path: (g722|ulaw) -> (g723)

Hello!

I installed the latest version of FreePBX (FreePBX 15, Linux 7.6, Asterisk 16), this error occurs during an incoming call:

[2020-04-22 10:04:12] WARNING[43734][C-00000046]: channel.c:5688 set_format: Unable to find a codec translation path: (slin) -> (g723)
[2020-04-22 10:04:12] WARNING[43734][C-00000046]: file.c:1262 ast_streamfile: Unable to open custom/Itirus-8khz (format (g723)): Function not implemented
[2020-04-22 10:04:12] WARNING[43734][C-00000046]: pbx_builtins.c:1175 pbx_builtin_background: ast_streamfile failed on PJSIP/anonymous-00000046 for custom/Itirus-8khz

[2020-04-22 10:04:22] WARNING[43734][C-00000046]: channel.c:5688 set_format: Unable to find a codec translation path: (g722|ulaw) -> (g723)
[2020-04-22 10:04:22] WARNING[43734][C-00000046]: file.c:1262 ast_streamfile: Unable to open no-valid-responce-pls-try-again (format (g723)): Function not implemented
[2020-04-22 10:04:22] WARNING[43734][C-00000046]: pbx_builtins.c:1175 pbx_builtin_background: ast_streamfile failed on PJSIP/anonymous-00000046 for no-valid-responce-pls-try-again

[2020-04-22 08:12:51] WARNING[22107][C-00000016]: chan_pjsip.c:959 chan_pjsip_write_stream: Channel PJSIP/anonymous-00000015 asked to send ulaw frame when native formats are (g723) (rd:g723->g723; wr:slin->ulaw;([email protected])->([email protected]))

itirus*CLI> core show file formats
Format Name Extensions


slin mp3 mp3
slin ogg_vorbis ogg
gsm wav49 WAV|wav49
adpcm vox vox
slin192 sln192 sln192
slin96 sln96 sln96
slin48 sln48 sln48
slin44 sln44 sln44
slin32 sln32 sln32
slin24 sln24 sln24
slin16 sln16 sln16
slin12 sln12 sln12
slin sln sln|raw
siren7 siren7 siren7
siren14 siren14 siren14
speex32 ogg_speex32 spx32
speex16 ogg_speex16 spx16
speex ogg_speex spx
slin48 ogg_opus opus
ilbc iLBC ilbc
h264 h264 h264
h263 h263 h263
gsm gsm gsm
g729 g729 g729
g726 g726-16 g726-16
g726 g726-24 g726-24
g726 g726-32 g726-32
g726 g726-40 g726-40
g723 g723sf g723|g723sf
g719 g719 g719
g722 g722 g722
ulaw au au
alaw alaw alaw|al|alw
ulaw pcm pcm|ulaw|ul|mu|ulw
slin16 wav16 wav16
slin wav wav
36 file formats registered.

itirus*CLI> core show codecs
ID TYPE NAME FORMAT DESCRIPTION

  31 image png          png              (PNG Image)
   6 audio g726         g726             (G.726 RFC3551)
   4 audio alaw         alaw             (G.711 a-law)
   2 audio g723         g723             (G.723.1)
  20 audio speex        speex            (SpeeX)
  21 audio speex        speex16          (SpeeX 16khz)
  22 audio speex        speex32          (SpeeX 32khz)
  24 audio g722         g722             (G722)
  25 audio siren7       siren7           (ITU G.722.1 (Siren7, licensed from Polycom))
  32 video h261         h261             (H.261 video)
  33 video h263         h263             (H.263 video)
   8 audio adpcm        adpcm            (Dialogic ADPCM)
  36 video h265         h265             (H.265 video)
  44 audio silk         silk8            (SILK Codec (8 KHz))
  45 audio silk         silk12           (SILK Codec (12 KHz))
  46 audio silk         silk16           (SILK Codec (16 KHz))
  47 audio silk         silk24           (SILK Codec (24 KHz))
  28 audio g719         g719             (ITU G.719)
  34 video h263p        h263p            (H.263+ video)
  35 video h264         h264             (H.264 video)
  19 audio g729         g729             (G.729A)
   9 audio slin         slin             (16 bit Signed Linear PCM)
  10 audio slin         slin12           (16 bit Signed Linear PCM (12kHz))
  11 audio slin         slin16           (16 bit Signed Linear PCM (16kHz))
  12 audio slin         slin24           (16 bit Signed Linear PCM (24kHz))
  13 audio slin         slin32           (16 bit Signed Linear PCM (32kHz))
  14 audio slin         slin44           (16 bit Signed Linear PCM (44kHz))
  15 audio slin         slin48           (16 bit Signed Linear PCM (48kHz))
  16 audio slin         slin96           (16 bit Signed Linear PCM (96kHz))
  17 audio slin         slin192          (16 bit Signed Linear PCM (192kHz))
   3 audio ulaw         ulaw             (G.711 u-law)
  18 audio lpc10        lpc10            (LPC10)
  27 audio testlaw      testlaw          (G.711 test-law)
  43 audio none         none             (<Null> codec)
  42 image t38          t38              (T.38 UDPTL Fax)
  39 video vp9          vp9              (VP9 video)
  38 video vp8          vp8              (VP8 video)
   5 audio gsm          gsm              (GSM)
  37 video mpeg4        mpeg4            (MPEG4 video)
  23 audio ilbc         ilbc             (iLBC)
  40 text  red          red              (T.140 Realtime Text with redundancy)
  41 text  t140         t140             (Passthrough T.140 Realtime Text)
  29 audio opus         opus             (Opus Codec)
  30 image jpeg         jpeg             (JPEG image)
   7 audio g726aal2     g726aal2         (G.726 AAL2)
   1 audio codec2       codec2           (Codec 2)
  26 audio siren14      siren14          (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))

I have never set up FreePBX before, please help solve the problem.

Thanks.

you have a codec mismatch , core show translations would be helpful

in general you need to have a codec enabled on the system able to negotiate with the incoming invite

check out what you configured in sip settings and on the trunk compared to what the ITSP or whomever is sending you the call expects wrt to codec

Where to check it all? This is the first time I’m setting up FreePBX and I don’t know much.

There’s a lot to unpack on that line:

  1. You should not have Anonymous Calling allowed on your system. This is the first route to toll-fraud and could end up with huge charges.
  2. Assuming you want that call, how did the connection get negotiated with G723?

If looks like you have some serious problems with your config.

First - turn off Anonymous Calling and Guest Calling before you get hurt.
Second - set up your trunk to this connection so that it only allows valid codecs. In general, you should stick to uLaw or aLaw (depending on your location) to start and then add the additional codecs once the rest of the issues are resolved.

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