I've been using my VPS (Asterisk+FreePBX) for a few years now, it proxies all my audio since it's on a public static IP. Never had issues. Recently I've been making calls to the DC/NoVA area and noticed that a few of my calls are having issues, the other side can't hear me nor does my DTMF make it through (i.e. automated systems don't respond).
I checked and saw that these troublesome calls were going through "Verizon (12)" in my AnveoDirect Smart Route. Blocked "Verizon (12)" for all US calls and voila, everything works normally. Now I'm trying to debug where the problem may lie.
After going through the asterisk command-line, RTP packets are being sent/received normally, even the DTMF packets. The only obvious difference I found was the following in the no-outgoing-audio calls, it shows up every now and then during the fast scrolling of log messages:
[2017-01-21 13:34:30] DEBUG[1100][C-0000000f]: res_rtp_asterisk.c:3893 process_cn_rfc3389: - RTP 3389 Comfort noise event: Format ulaw (len = 1)[2017-01-21 13:34:30] WARNING[1098][C-0000000f]: chan_sip.c:7496 sip_write: Can't send 10 type frames with SIP write
Funnily enough, I can't remember which numbers caused this issue except for one of them. Washington Gas (17037501000), their automated system won't recognize any DTMF, that's how I've been testing it. I did go to customer service a couple months ago and they couldn't hear me, had to call back using my cell.
The WARNING only shows up with the above phone number + Verizon (12) combo, change any of those two things, and no issues. I don't know if that's the cause or just a symptom of the real cause. Any clue as to what the problem may be? Me? Verizon? AnveoDirect?
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