ASTERISK 20 Codec issue

Asterisk Asterisk Support
1

[Aug 2 09:23:10] WARNING[142355][C-00000001]: translate.c:494 ast_translator_build_path: No translator path: (ending codec is not valid)
[Aug 2 09:23:10] WARNING[142355][C-00000001]: translate.c:494 ast_translator_build_path: No translator path: (ending codec is not valid)
[Aug 2 09:23:10] WARNING[142355][C-00000001]: channel.c:5765 set_format: Unable to find a codec translation path: (g729) → (ulaw|alaw|gsm)[Aug 2 09:23:10] WARNING[142355][C-00000001]: app_dial.c:1773 wait_for_answer: Unable to write frametype:

how do i fix this

2

The g729 codec is in use, and there is no included translator for it. Either switch away from using g729, or acquire a g729 codec module.

3

how can i do it

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5

and if you wand to be cheep and know how to code and do your own debugging
and ignore licens fees

6

but if you want to just fix you problem change pjsip.conf

allow = !all,g722,ulaw,alaw
7

actually am not able to hear the one am calling i need to fix it the trunk provider said add g729 and i use pjsip also the icon in micro sip at left bottom is red when i answer the call

8

can you post the INVITE / 200 OK from the provider (we need the SDP part)
i have never heard about any provider that do not support alaw or ulaw
as Asterisk do not support g.729 without change you should try the allow line i posted

9

G.729 DOES sacrifice call quality! It has a lower Mean Opinion Score of 3.75 to 3.8 as against 4.3 to 4.4 for G.711. It is not suitable for music.

I seem to remember that it is no longer the best low internet bandwidth codec.

10

This doesn’t sound like a “codec” issue…No pun intended. It sounds more like a NAT issue

11

It’s not your “trunk” provider if you can’t hear the other person on the line. It’s a nat issue / firewall issue from your end.

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