similar to: FXO Problems

I tried using CVS HEAD 08-05-2004 for zaptel/asterisk last night but I'm having problems with zaptel. The last CVS HEAD I was using was CVS HEAD 07-31-2004. (BTW... I'm a CVS moron... what command would I use to get back to that version of HEAD because like an idiot I forgot to make a backup copy of the CVS HEAD on my system before I install from the new CVS HEAD. I tried cvs co -D

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

Location = US asterisk/zaptel from CVS. Updated last week some time. Currently rebuilding with todays checkout. I have 2 fxo channels hooked up to outside standard Bell South phone lines. If I configure as so [channels] context=pstn group = 1 signalling = fxs_ks callprogress = yes channel => 4,3 Then any call routed from asterisk to the outside line will ring, and can be picked up, but *

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

I have some X100P connected to my analog PBX. When I want to call an analog extension on that PBX I use the following rule: exten => _21XX,1,Dial(Zap/g1/${EXTEN:2},20) where 21 is just a prefix to indicate it?s an analog extension and XX matches the real two digit extension number. (this is why I strip of two digits when dialing Zap/g1. Well, everything works fine, except that * says on

Details: Asterisk 1.0.9 Zaptel 1.0 Dell P3 1ghz with X100P Clone Location: India This is an interesting issue where when I open up ZTMonitor, it shows the RX as being on. It seems that Zaptel doesn't know to hang up the line so after a couple of hours when the telecom cuts the line, everythign stops working. Things I've tried include playing with the zaptel.conf, trying zaptel v1.2(with

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'm having some trouble dialing phone numbers in Argentina with Digium Zaptel cards. Does anyone have some sample configuration that works with Digium TDM04B cards in Argentina? I'm mainly referring to the /etc/zaptel.conf and /etc/asterisk/zapata.conf. I have two Zaptel cards: the first one has 1 FXS port and 3 FXO ports; the second one has 4 FXO ports. My current configuration is

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Hello, I am getting problem for delay call hang-up with the below scenario: PSTN User (calling Party)------------->PSTN Line ----------> FXO with Asterisk Box----------->SIP IP Phone (called party) I am using X100P card with my Asterisk-1.0.7 box. I am also using Zaptel-1.0.7 version. When PSTN user makes call to my PSTN line and after getting IVR, PSTN user dial my SIP

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

Hello, I've got an Asterisk server with a TDM400P with 2 FXO modules and 2 FXS modules. This server is connected to 2 PSTN lines and 2 analog phones. In my Zaptel configuration, I've defined 2 groups : one for the 2 FXO's and one for the 2 FXS. The asterisk server is just used to add a little IVR and Voicemail service. Eveything works fine, but sometimes the conversation is

Hi... I have the next problem.. I have a FXO card with i can make calls but i cant recive calls. At the consol, i get the next error: -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Attempting native bridge of Zap/1-1 and Zap/2-1 WARNING[262160]: File chan_zap.c, Line 2857 (zt_handle_event): Ring/Off-hook in strange state 6 on channel

I have an eight channel fxo setup (2 TDM400P cards) and I have them setup. Here are my configs: Zaptel.conf: fxsks=1-8 loadzone=us defaultzone=us Zapata.conf: [trunkgroups] [channels] musiconhold=default rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callerid=xxxxxx callwaiting=yes busydetect=no callprogress=no usecallingpres=yes callwaitingcallerid=yes

so i have been running v1-0 on all of my test boxes for about a month now testing iax/sip/res_xxx. I decided to put it into production so I updated a box that was running 0.9.? that had been working perfectly for months and low and behold the inbound line from telco now intermittantly doesn't clear and none of the other channels can dial out on that line. I have tested the line in this

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes