I am having numerous problems with my SPA-8800. I am using the SPA-8800 as an FXO gateway for 4 verizon fios analog trunks into a PBX In A Flash (Asterisk 1.4) system. I am not using any of the FXS ports. The FXO ports are used for all inbound and outbound voice traffic into and out of the Asterisk system. I am having the following problems, in order of annoyance:
1. about 20-30% of the time, when somebody dials out (using the single set of configured outbound routes), they will hear a second dial tone - instead of the call going out, something is returning a second, unbreakable dialtone. Based on the syslog messages, it looks like the SPA-8800 is getting the request from the Asterisk, but somehow, it is not completing the call. If the calling party hangs up, and then redials, most times the call will complete (on the same trunk that it failed on earlier). Sometimes they will have to try multiple times. Not sure how to troubleshoot this - I have tried to change the PSTN Dialing Delay but still seem to have the problem. Inbound seems to be fine most of the time, although there are times when I have had complaints that people have called and it rang with no answer - which indicates the calls where not ending up on the asterisk. Haven't had the complaint in a while, since I moved the firealarm system off my main incoming numbers.
2. Echo - I have local echo (i.e. the Asterisk user) hears echo - I have got it down to a fainter echo - but it is always seems to be there - just tried decreasing the SPA to PSTN gain another DB (from -6 to -7) to see if this helps. I have no echo issues on station to station calls, only on outside trunk calls. The party on the PSTN side of the call has no echo or other audio issues. I have tried everything in the Reducing the Echo on the PSTN line of SPA3102 document off of the Cisco web site. It seems to have helped, but I still wonder if there is something I am missing ...
That's it - hopefully somebody out there can shed some light on these problems ...
I have attached an HTM from the SPA-8800. Here is the asterisk config on one of the trunks (they are all identical with the exception of the port, username and secret:
This trunk's name is pstn2, the PEER info is as follows (with some comments):
host=192.168.1.XXX <- IP of the SPA-8800 username=XXXXXXXXXX secret=XXXXXXXXXXXXX type=friend port=5161 <- this would be line 2 dtmfmode=rfc2833 insecure=port,invite allow=ulaw canreinvite=no nat=no call-limit=1
Thanks for the reply. I think I solved problem 1 (or at least minimized it). Just before sending out the HTM file last week, I changed the DTMF level from -16 to -10 - that seems to have solved the intermittent second dial tone. This morning I changed it back and immediately started running into the second dial tone problem - so I think that this is the problem: The 8800 was generating DTMF that was on the border of being too soft for our trunks.
I still have some echo issues - I may try lowering the VOIP to PSTN level another db to see if this helps. I am already down to -7.
Unfortunately, I was never able to get this to work right - always had too much echo on the lines. I ended up buying a digium TDM410 clone off of ebay for 84.95 (including shipping). It works much much better than the spa-8800 - no echo problems.
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