similar to: TDM11B FXS port stops working after a reload?

Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried

Hi list, My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get this error message on the asterisk console: == Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' [Jul 8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf: Couldn't detect start-character. CID parsing might be unreliable A long time ago my

Hi all... I have a problem with caller id on my asterisk server. here is my configuration : centos-5, asterisk 1.6.2.1, dahdi-linux-complete-2.2.1, libpri-1.4.10.2 ibm X-3200 series, xorcom astribank (16fxo, 8fxs), 16 line telco (hunting) everything fine until I try to feed my app with caller id. My extensions.conf : [incoming1] exten =>

Everyone, Im sure you've seen this error a million times, but Ive looked everywhere I can think of & still haven't found a solution that works. I'm trying to make an outside call, I can call the physical phone from a xlite on another pc (and vice versa) but whenever I try to make a call to the outside world, this happens: on the CLI: Jun 15 08:45:20 NOTICE[10390]:

Hi all! I am in the same situation here in Sweden, but I just got the X100 card and don t get it to work with DTMF caller ID. Is the only way to solve this to buy a TDM400 card w FXO modules? Is it just a driver question or a HW issue with the X100 card? Where to find dok about the new keywords in conf files like "cidstart=polarity" in zapata.conf? rgds Gunnar >>>

I had to return my TDM11B because it put the PSTN line 'off hook' the moment I plugged it in and wouldn't hang it up. The new card seems to work because I can actually make an outgoing call from the FXO port to my cell phone, so I'm pretty happy about that. But Asterisk doesn't recognize incoming calls from the PSTN. If I dial my home phone from my cell phone asterisk

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

I'm testing a TDM400P with FXO module to receive incoming calls from an analogue line and send it to a SIP device. To recieve callerid, I need to use cidsignalling=dtmf and cidstart=polarity. The problem is that when a call is finished, the TDM400P seems to require about 20 seconds to prepare for the next incoming call. If a new call comes in within 20 seconds after the previous call was

I've updated to the latest CVS as of today (and rebuilt RedHat 9). My setup is as follows: Wildcard X101P - channel 1 TDM400P - channels 2-3 - fxs cards with fxo signalling, channels 4-5 - fxo cards with fxs signalling I got CallerID to work on channel 4 with an old CVS, despite the usual "Didn't finish CallerID spill" message. However, as soon as I insert the following

Hi! I'm trying to set up an asterisk based PBX with a TDM400P +2 FXS +2 FXO modules in UAE/Dubai for home switching / voicemailing. I am using the card Asterisk/Zaptel 1.4.0. I want to include a special route when a certain caller calls into via PSTN. The problem is that I cannot detect the Caller ID. I tryed various setting (cidsignalling, cidstart) in my zapata.conf, here is the last

Hey all, I recently noticed that DTMF caller-ID was implemented in CVS, so I requested the service from my telco (the Dutch KPN) and tried to get it going in Asterisk (current CVS), without success so far. This system has 1 X100P, 2 TDM400P's with 4 FXS-modules each and 2 HFC-PCI ISDN-cards (zaphfc-driver) in it. The analog line I'm trying to get caller-ID working on is obviously on the

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

(Sorry of this appears in the list twice, but I wasn't sure if it was blocked or not) Hi there, I'm having a problem with my TDM-400P which has been working like a charm up until very recently. It started to fail last week, and so I was hoping someone could illuminate me with some information as to why. Its configuration is as follows: ------------ FXS (green) module is in position 1,

Hi, I got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

I recently purchased a TDM11B so I could hopefully hook flash the FXO from either the FXS (on the TDM11B) or a SIP device. From the FXS, I've tried hitting # then transferring to an extension that flashes the line then dials the FXS again (3020). This seems to send me to a busy signal and the console tells me no such host of 3020 (the number I'm on). The call on call waiting gets sent

Hello, my config : aah 2.6 (asterisk 1.2.4) , centos 4.2, 1 TDM11B (1 Fxo / 1 fxs ), 1 phone, 1 softphone I'm in France When someone from PSTN calls and hangs up before the call is answered, internal extension keeps ringing until timeout occurs. PSTN line keeps busy. Hangup detection doesn't work. I've played with different paremeters (callprogress, busydetect, busycount,

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

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, I am using asterisknow 1.5.0 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. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.