similar to: Caller ID on Asterisk and Astribank

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

Hi all... I'm sorry for repeating my message. I have a problem with caller id on my asterisk server with xorcom astribank. 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. here is the log : --

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

Hello, I've one astribank with 8 FXO unit and 8 pstn lines connected to the astribank. When I receive calls on my ipphone I get always Unknown callerid. It's is possible to receive the callerid from the lines on the astribank unit? This is my config: [channels] language=es context=from-zaptel signalling=fxs_ks ;rxwink=300 usecallerid=yes callerid=asreceived ;cidsignalling=bell

Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [s at from-pstn:3] Hangup("Zap/5-1", "") in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17

Okay, I can't find what might be causing this. Here is what I got: Asterisk server hooked up to a digital T1 line (full 24-channel) via a Digium card. Verizon has turned on caller ID on the first line (I can guarantee it is on as I can hear the FSK tones on this line but not the others). Using zttool an ZapScan() I have determined the following: 1) The RxB/RxD bits toggle from 1 to 0

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.

<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

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

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

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

Hello gang, We just got MaBell to turn on our callerid. I tested the capability with a southwest bell box and a plain phone, so I know the line is sending the signal. I'm running Asterisk SVN-branch-1.4-r204834 using a TDM400P card. Here is my dahdi_cfg -vv output: dahdi_cfg -vv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.3 Echo Canceller(s): MG2

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.

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 >>>

Aloha, I'm finishing up the final touches on this install, and have run into an odd problem. I can't seem to get Caller ID on the analog phone lines working. It's a Digium AEX 410 card. I have Verbose set and a line to print the CID: I have usecallerid=yes and callerid=asreceived set in both chan_dahdi.conf and users.conf [analog] include=>default exten =>

Hi list First of all could you please explain loop-start and ground-start for me? What are they used for? Next, I have the following configurations: dahdi-channels.conf : context=pstn-channels signalling=fxs_ks channel=>130 context=phone-channels signalling=fxo_ks channel=>127 chan_dahdi.conf : [channels] cidsignalling=dtmf cidstart=dtmf signalling=fxo_ls pulsedial=no

I have a pretty large setup on one of my customers. Digium TE420B (with echo cancelling module), 3 Xorcom Astribanks with 32 FXS each and 1 Xorcom Astribank with 16 FXO. These FXO ports are NOT used for fax/data transmission, as they are connected to cell phones. Not really related to the issue, but there are also 250 SIP phones. The problem is that fax and dial-up connections are really

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

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

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