[Asterisk-Users] FXO Problems

[Mike Coakley]

I talked to Digium this afternoon and it appears that they believe the 
problem is with my cards. I received them around June 21st and they 
have apparently released a modification around or after that time. (The 
Digium technician had me look for a mod-wire on the back of the card, 
which was not present.) Unfortunately, I could not get the card 
revision for him but he felt sufficiently sure that the problem was 
with the card so we are RMAing the cards. When I take the cards out I 
will notify the list with the revision # so other people can compare 
with their cards. (I have the TDM400P w/4 FXO bundle - I didn't 
originally mention that specifically.)

Mike

On Aug 6, 2004, at 2:27 PM, Mike Coakley wrote:

> I have 2 Digium 4 port FXO cards in my system. The system is a P4 
> 2.4Ghz, 512MB RAM, Promise FastTrax 100 TX2 Pro Raid, 80GB RAID1  for 
> storage - whitebox - running RedHat 9. With pretty much any CVS HEAD 
> version we are getting, what I will call, "phantom" calls on some 
> lines. What I mean by a phantom call is that the line will ring, 
> Asterisk will log that the Zap channel has been answered, the context 
> will call all of our SIP phones (which works fine) but when you pickup 
> the handset you get a dial tone. If you just sit there and listen 
> Asterisk will log that the Zap channel has hung up (after about 5 
> seconds) and the SIP phone goes busy. I am getting a "reverse 
> polarity" message on the console of the Asterisk system (not in the 
> Asterisk console but on the monitor attached to the hardware). But I'm 
> not sure that is even related yet as today is the first time I saw 
> this message.
>
> I have played with the busydetect and callprogress settings but 
> nothing has helped. (So I have left them off for now. With each new 
> CVS HEAD I download I try all the settings again but they have not 
> changed the situation.) I should also say that this problem did not 
> exist before I switched over to the Asterisk system. The key system we 
> had before did not exhibit this problem. SO I am assuming it is (a) 
> the new wiring we put in place to get to the Asterisk FXO cards (but I 
> have replaced that during my diagnosis already), (b) the way that the 
> Digium cards are handling the teclo terminations, or (c) my 
> zaptel.conf or zapata.conf are somehow wrong.
>
> Here are my configs:
>
> ZAPTEL.CONF
> #
> # Zaptel Configuration File
> #
> fxsks=1-8
> loadzone = us
> defaultzone=us
>
> ZAPATA.CONF
> ;
> ; Zapata telephony interface
> ;
> ; Configuration file
>
> [trunkgroups]
>
> [channels]
> ;
> ; All channel defaults
> ;
> language=en
> context=pstn_inbound
> signalling=fxs_ks
> musiconhold=default
>
> relaxdtmf=no
> echotraining=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=5.0
> txgain=0.0
> busydetect=no
> busycount=6
> callprogress=no
>
> callgroup=1
> pickupgroup=1
>
> ;We have this disabled for now since we don't have Caller ID on our 
> lines
> usecallerid=no
> useincomingcalleridonzaptransfer=yes
>
> ;We group the following lines into a single group for
> ; pooled outbound calls
> group=1
>
> ;Define our incoming/outgoing lines
> callerid="MB Main - Line 1" <(973) 252-xxxx>
> channel => 1
> callerid="MB - Line 2" <(973) 252-xxxx>
> channel => 2
> callerid="MB - Line 3" <(973) 252-xxxx>
> channel => 3
> callerid="MB - Line 4" <(973) 252-xxxx>
> channel => 4
> callerid="MB - Line 5" <(973) 252-xxxx>
> channel => 5
> callerid="MB - Line 6" <(973) 252-xxxx>
> context=pstn_fax_inbound
> channel => 6
>
> I'm sure there are some telco gurus out there that can tell me how to 
> test the incoming telco lines to determine where my problem is. My LEC 
> is Verizon in New Jersey for those who care.
>
> Some more background: I have had other problems on the line. I had a 
> beeping or blip on the line towards the caller (not the receiving SIP 
> side) that sounded like the blips you would get if you were recording 
> the call. Reading some other posts it sounded like another problem 
> someone else was having and giving the Digium FXO cards a higher PCI 
> latency timing helped that situation. Also I had echo on the lines and 
> no settings in the configs helped at all. I had to uncomment the 
> AGRESSIVE_SUPRESSOR for the MARK2 echo cancellation algorithm to get 
> my echo to go away (thanks Mark from Digium for that one).
>
> Thanks,
>
> Mike
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>