I’m really hoping that this is a dumb newbie question so I can be laughed at and set on the right track, but I’ve been getting nowhere with it for days now, so it’s time to ask.
I have two trunks, BT via DAHDI and SIP via voipfone.co.uk
I have SIP (Linphone) and DAHDI (analogue Panasonic and BT handsets) extensions.
For DAHDI the system has a TDM400P in it.
Everything was working fine. It was trouble free, I was getting on with other things. Then, my problems with echo started. I hadn’t changed anything on the Asterisk box just before this happened. I waited for a while in the hope it would just go away again, but it hasn’t. The system is very lightly loaded, in fact less so than it was before. There have been a couple of hardware changes, but these don’t coincide precisely with the problems and I have checked that there isn’t anything else using the TDM400P’s IRQ.
The problem is intermitted, and its severity waxes and wanes, but the general trend seems to be as follows.
Calls between the SIP trunk and a DAHDI extension frequently have a painful echo on the remote end; the external party speaking over the trunk hears themself coming back at full volume with a good second’s worth of delay.
Calls between the DAHDI trunk and a DAHDI extension produce a local echo on the DAHDI extension, at full volume, but with practically no delay. The trunk end is fine.
Calls between the SIP trunk and a SIP extension (computer mic and speakers - a recipe for feedback I know, so the least reliable test, but interesting because it was fine before) produce a fainter echo for the external party with about 500ms delay. If the computer’s speakers are switched off, the echo stops (as I would expect).
Calls between the DAHDI trunk and a SIP extension produce a very mild echo on the both sides with a tiny delay which isn’t far off what I’d expect naturally from using the mic and speaker softphone setup. This may be nothing.
Calls between SIP extensions - hard to tell because the quality on one of them has gone awful (I think it’s a software issue with the client), but sounds like a mild echo both ends.
Calls between DAHDI extensions produce an almost instant loud echo at both ends.
Calls between a SIP and a DAHDI extension produce an echo at the SIP end of about 250ms and a virtually instant one at the DAHDI end.
Calls between either SIP or DAHDI trunk or extension and Asterisk voicemail produce no echo.
At the moment, my /etc/asterisk/dahdi-channels.conf has a single echocancel=yes at the end of each channel definition. These weren’t there before - I’ve added them since the problems started, but they don’t seem to have made a difference.
; Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) ;;; line="1 WCTDM/4/0 FXOKS (In use) (EC: OSLEC - INACTIVE)" signalling=fxo_ks callerid="Channel 1" <1231> mailbox=1101 group=5 context=from-internal faxdetect=incoming channel => 1 callerid= mailbox= group= context=default echocancel=yes ;;; line="2 WCTDM/4/1 FXOKS (In use) (EC: OSLEC - INACTIVE)" signalling=fxo_ks callerid="Channel 2" <1232> mailbox=1102 group=5 context=from-internal channel => 2 callerid= mailbox= group= context=default echocancel=yes ;;; line="3 WCTDM/4/2 FXOKS (In use) (EC: OSLEC - INACTIVE)" signalling=fxo_ks callerid="Channel 3" <1233> mailbox=1103 group=5 context=from-internal channel => 3 callerid= mailbox= group= context=default echocancel=yes ;;; line="4 WCTDM/4/3 FXSKS (In use) (EC: OSLEC - INACTIVE)" signalling=fxs_ks callerid=asreceived group=0 context=from-analog faxdetect=incoming channel => 4 callerid= group= context=default echocancel=yes
I’ve read loads of documentation on the causes of echo, but nothing seems to stack up very well with what I’m finding. That may be my lack of experience, though. Does this scenario immediately suggest anything to someone more clued-up? Are there other tests I should do or things I should look at to try to debug and/or solve this?