similar to: Echo / Adit 600

I have an adit 600 with one fxo card connected to a Digium single span T1 card. CallerID, disconnect supervision work perfect, however the users complain that they have some sound quality issues, after testing it I realized that whenever one is in a phone call they get like silence between the sounds coming from the other party, almost like a cell phone, in other words if there is no sound coming

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

Hello all! I have tried and tried to resolve this one to no avail. Hopefully one of you can help... The system in question is a Compaq Evo D600 (iirc) business desktop, with a 1.4GHz Pentium 4 and 512mb of RAM, running a stock install of PoundKey 1.2. It has two Digium cards installed: a TDM400P with four FXO modules, and a TE110P hooked to a Carrier Access Adit 600 which serves 8

I'm a bit confused on how to setup Zaptel.conf and Zapata.conf when there is a TDM400P and a TE410P installed after upgrade. The TDM400P has 2 FXS in position 1 & 2 and 1 FXO in the fourth position. I see boot, WCT4xxP loading first and WCFXS loading second. According to my understanding, given above, the TE410P should be configured first, then the TDM400P. However, I'm not sure

I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

Hi ! I am connecting to Adit 600 thru a T100P card I have configured 1-16 FXS channels and 17-24 FXO. Everything looks fine on Asterisk side I get a tone on all FXS channels, but when I try to dialout thru one of the FXO channels 17-24 it doesn't connect to the POTS line and echoes back my voice. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit

Hi, I've connected a two T100P from digium with a 2 Rhino channelBank. Everything is working as expected. but I have occasional Falls, asterisk take 99% of CPU resources, with the following report May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n** May 24 14:52:41 WARNING[11441]: We're Zap/1-1, not H8?F??n**

When I enable callprogress I get this error message... (when I call, it will ring forever but asterisk is acting as if it DID pick up the line ... but it never did... ) I'm beginning to think that asterisk can't hangup the line when in voice mail and this, whatever action I'll take... Actually, callprogress doesn't seem to work at all... (in my case) Mar 24 23:37:28

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

Hi I am having problems getting my card to hang up properly when a remote party hangs up the line. I know i have to use the busydetect stuff but it doesn't seem to be working. It is a BT line and my zapata.conf is as follows: [channels] language=en rxgain=0.0 txgain=0.0 immediate=no echocancel=yes echocancelwhenbridged=yes echotraining=yes signalling=fxo_ks context = dialphone channel =>

Hi everyone, Sometimes i am having problems with Zap channels on asterisk 1.2 (Disc-OS 1.1), after some calls, the channel continues in use, even after hanging the call up, then i need to run the "soft hangup Zap/<zapchannel>" in the asterisk CLI to release the channel. Here is my zapata.conf: [trunkgroups] [channels] language=pt_BR context=default usecallerid=yes

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured

I am a little confused about the FXO ports on the Carrier Accesss Bank 1. The wiki, and people on the mailing list have stated that they do not have impedance matching, but Carrier Access states in their documentation that they do have impednace matching. Did Carrier Access make a typo, or does it depend on the version of the FXO card? I current have a CAC AB1, and I am having echo problems

For some reason, I can't get call pickup to work between Sip phones or between Sip and Zap phones. All phones are in the same call group and pickup group (1). The source code was downloaded and built as of today 11/15/03. Here's what's in sip.conf: [general] port=5060 bindaddr=192.168.17.2 tos=lowdelay disallow=all allow=ulaw context=aliens ; ; SIP Entry for sipura line 1 ; This

Hey everybody, I've had issues with detecting call disconnects on the Centrex system and an Adit 600. I've discovered that if I set the channel bank's FXO to lscpd, even though the CO says they don't support, that Asterisk can see the hangups. The catch? Asterisk can no longer seize the line. I've used this to my advantage, since I have 13 lines to work with. (5) +1

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1