similar to: Bug ???

<font size="2">I've got an odd situation where callerid is only picked up every other call.&nbsp; Is there anything I can do so callerid works on all calls?<br><br>I'm seeing the channel hangs up during a call<br><br>&nbsp; == Starting post polarity CID detection on channel 4<br>&nbsp;&nbsp;&nbsp; -- Starting simple switch

Hello list. I have a question about how to read the incoming calls' callerid on an FXO interface of a TDM 400 analog card; (it's one of those RED modules). Now -may this is the complexity adding step..- I have a GSM gateway attached to this FXO thing; incoming calls are processed as they should. But both when peeking on the CLI, as well as in the phone display I do not see the caller id.

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up "from" my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking

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

So I have an application where the users want to divert incoming calls on one analogue line out to another analogue line - both lines are supplied by BT and theres a TDM400 in the box. Call comes in, system Dial's the forwarding number and bridges the calls. Works fine. Until one (or both) leg hangs up. And the box never sees the hangup. So it sits there with both Zap channels open

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, I'm just starting out with Asterisk, in combination with a TDM400, filled with 2 FXS on channels 1 and 2, and 1 FXO on 4. Having just started, all I want right now is to be able to answer incoming calls on a phone connected to channel 1. The trouble is the caller id. I have caller id enabled on my line, my phone supports it, and when I connect the phone directly to the line, it works.

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 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

Dear All, I am a newbie about asterisk. I have 1x X100P card 3x Sip phone I got aware of problem, after I saw the caller id on my sip phone. I noticed that if I receive a call from GSM Operator A, I can see caller id. But any other operator, I got no caller id, even my direct PSTN service operator. So at that moment I was using *1.0.9. than I changed to asterisk@home 1.3(1.0.9). I got same

Hello, I have recently purchased a TDM400P card with one FXO expansion card, and I'm having problems. The card does not pick up incoming calls. Asterisk detects the ringing line and rings various SIP phones as required. When a sip phone answers, the sip user hears nothing and the PSTN user continues to hear ringing. Here is the asterisk output for an incoming call:

Hi, You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy. Idris _____ From: phil.dawson@marnock.com [mailto:phil.dawson@marnock.com] Sent: Wednesday, June 21, 2006 12:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Monitor a particular SIP call for training purposes Hi, I've been asked if it is possible to allow a user to

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

I am trying to reconfigure an asterisk box that was using an HFC-S card with bristuff but is now using 2 analog lines therefore I want to use the TDM02B to connect to two POTS lines. The TDM02B has 2 red modules. I have this in /etc/zaptel.conf loadzone=nl defaultzone=nl fxsks=1-2 I have /etc/asterisk/zapata.conf signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=400

supervision. Verify if for those "numbers" the CO revert the line polarity when callee answer. callprogress=no is a good test too. Jorge Ra??l G??mez C. wrote: > Hi list, > > I'm having problems transferring certain calls made by the attendant > between the PSTN and to an internal extension. Although, transfers > between the majority of the calls ends successfully.

Salve Salve Galera. Tenho a seguinte situacao: Uma placa MD3200 ligada em uma linha telefonica comum(PTSN) e funcionando "belezinha"... Tenho configurado um URA, onde ele atende a ligacao que chegou no canal e solicita o numero do ramal de destino da liga??o: Acontece que ao discar o ramal de destino, ele nao encaminha a ligacao, ficando mudo e posteriormente caindo a liga?ao. Fiz

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)

Hi list, I'm wondering if it's possible to transfer a call that is still ringing??? Actually, the problem is that my telco provider doesn't offer an uniform method for answer/disconnection supervision, and by that I mean, some of it's line (I think) offer a polarity reversal, but other lines (of the same service provider) do not offer anything at all, so the answer of a call

Hello, I am trying to make my Asterisk to recognize CallerID from incoming PSTN calls. I am using a TDM400. CallerID in my country's PSTN is based on DTMF. Some information from my configuration files: /etc/zaptel.conf: loadzone=nl defaultzone=nl /etc/asterisk/zapata.conf usecallerid=yes cidsignalling=dtmf cidstart=polarity As displayed on the console, my Asterisk cannot detect the event