We are using FreePBX 13 with Asterisk 13.21 certified and started getting a lot of call quality issues. We are doing a 3-way conference using AMI originate on asterisk, everything works great but sometimes when we have a good bit of calls, audio getts so choppy and we believe asterisk or our OS starts dropping UDP packets.
We are using the ULAW period for all trunks and calls. And one thing we noticed that recently we started getting this error message on asterisk console ->
chan_sip.c:23007 func_header_read: This function can only be used on SIP channels.
And every time we get this message, our network traffic on asterisk spikes up for a few seconds. We are just using chan_sip not pjsip on this server.
Please help us to identify the cause of this issue. Any suggestions will be highly appreciated.