No translator Path

Asterisk Asterisk Support
1

Hi,
This is scrolling constantly in my logs. Is there any way to pin point why this is happening?

[Dec 13 16:24:16] WARNING[28619][C-001436be]: translate.c:487 ast_translator_build_path: No translator path: (ending codec is not valid)

Im using SIP and these codecs:

format_gsm.so Raw GSM data 0 Running core
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 43 Running core

2

format modules are not codecs. There aren’t actually modules for codecs, only for translations between them.

You can, for example, have a g729 format module, allowing you to play pre-recorded media in g.729 format, without being able to build a translation path to any other format.

However, I have a feeling that invalid means an unknown codec.

1 Like
3

any ideas how that could be triggered or how to fix the issue?

4

We probably need the protocol codec negotiation, or details of the selected codecs, if the channel type doesn’t negotiate them. Also, if this is an RTP based, technology, the format codes for all the RTP packets.

5

Honestly, I’m lost on how to get any of that.

In sip.conf I have disallow=all and allow=ulaw

The only rtp settings I have are:

rtpstart=10000
rtpend=20000

The modules I have loaded are:

app_chanspy.so
app_confbridge.so
app_controlplayback.so
app_dial.so
app_directory.so
app_echo.so
app_macro.so
app_mixmonitor.so
app_page.so
app_playback.so
app_senddtmf.so
app_stack.so
app_transfer.so
app_verbose.so
bridge_builtin_features.so
bridge_simple.so
bridge_softmix.so
chan_dahdi.so
chan_sip.so
codec_gsm.so
codec_ulaw.so
format_gsm.so
format_pcm.so
format_wav.so
func_callerid.so
func_channel.so
func_dialplan.so
func_jitterbuffer.so
func_periodic_hook.so
func_volume.so
pbx_config.so
pbx_spool.so
res_agi.so
res_ari.so
res_crypto.so
res_fax.so
res_fax_spandsp.so
res_monitor.so
res_musiconhold.so
res_rtp_asterisk.so
res_smdi.so
res_sorcery_config.so
res_speech.so
res_stasis.so
res_timing_timerfd.so
6

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information#CollectingDebugInformation-Enablechanneltechorfeaturespecificdebug