I am using Asterisk 20.4.0 and FreePBX along with wdoekes/asterisk-chan-dongle to handle calls on my SIM card via VoIP. The incoming audio from the dongle to my PJSIP extension is working perfectly. However, I’m encountering issues with outgoing audio from my PJSIP extension to the dongle, as it intermittently cuts in and out during calls, to a severe extent.
I have observed warnings like the following in the Asterisk console while on a call:
translate.c:603 ast_translate: 9540 lost frame(s) 9541/0 (slin@48000)->(opus@48000)
I attempted to address this issue by changing the audio codec in the SIP settings from
ALAW, but I’m still experiencing audio cuts and receiving similar warnings in the logs:
translate.c:603 ast_translate: 2861 lost frame(s) 2862/0 (slin@8000)->(ulaw@8000)
translate.c:603 ast_translate: 13543 lost frame(s) 13544/0 (slin@8000)->(alaw@8000)
I would greatly appreciate any insights or suggestions to resolve this issue.