I’m running Raspbx 10.10 (Asterisk 16 & FreePBX 15) on my RPi 4 with chan_dongle as a GSM gateway and whenever I place a call through the gateway I get no audio back and this message pops up in the console:
[2021-09-20 16:46:50] NOTICE[12384][C-00000002]: translate.c:603 ast_translate: 19066 lost frame(s) 19067/0 (slin@8000)->(ulaw@8000)
[2021-09-20 16:46:54] NOTICE[12384][C-00000002]: translate.c:603 ast_translate: 19278 lost frame(s) 19279/0 (slin@8000)->(ulaw@8000)
[2021-09-20 16:46:58] NOTICE[12384][C-00000002]: translate.c:603 ast_translate: 19500 lost frame(s) 19501/0 (slin@8000)->(ulaw@8000)
I already allowed all slin codecs in SIP settings but it still doesn’t work.
Could you provide the exact version as the contents of line 603 varies across the sub-versions of Asterisk 16, and I’ve been unable to find the code that is producing this exact message.
I’m not convinced this is a translation failure; it could mean there is a gap in the frame sequence numbers, but, in any case, the translations from slin will generally be done by the more specific codecs. You can use, I think, “core show translations” to see whether there is a translation path available.
Thanks for your reply. I kept searching yesterday for a solution and found the following post:
In the last comment there was den054 who had the same problem and issued the following AT command:
AT^SYSCFG=13,0,3FFFFFFF,0,3 - modem 2G only, automatic search any band, no roaming
This also worked for me!
According to Evgeniy Reshetnyak on the same post.
“One way audio problem during outgoing calls is because asterisk never get
ANSWER state from chan_dongle and always has pre-answer (early media) state.
May be it is incorrect processing AT response code from E3131?
I see the same problem is in Giovanni’s logs. http://code.google.com/p/asterisk-chan-dongle/issues/detail?id=129#c2”
I checked a bit more and found the whole list of arguments if you guys need it in the future: