similar to: DTMF CallerID

Good morning, I have a digium wctdm24xxp in my asterisk box, i am not able to see the callerid when the call is incoming from the fxo line, i live in Brazil, how can i change the signaling from fsk to dtmf? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I'm using Asterisk 1.4.18 (I've tried 1.4.19,1.4.21 too) and zaptel version 1.4.11. Card is "Digium Wildcard TDM800P", with driver wctdm24xxp. From Asterisk side this card has FXS ports, and FXO from outside. I've connect to them GSM-FXO gateway Benq C5 APC-868 (http://www.kontec.ru/c5.php). The problem is that this equipment sends caller id information in format

Using distinctive ring detection with bell202 cid, is there any way to tell DAHDI to sometimes expect the cid after the 2nd ring, other times after the 1st? I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P FXO line. No problem setting dringcontext for the 2nd DID. The 1st DID works normally, but I get no CallerID on the 2nd because the call is picked up before the

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

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

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

I have a TDM800P with 5 POTS lines hooked into my university's PBX. CID information is being picked up properly from phones inside the PBX, but not from phones outside. I've tried mucking with rxgain with no result, likewise cidstart and cidsignalling. The CID number shows up properly if a telephone with a CID display is plugged into one of the POTS line, whether the call is from inside

Hello, Talking with Soren Sratje about Caller ID in Brazil, we compare ours DTMF tones captured by ztmonitor. wcfxo correctly recognize the "DTMF CLIP" and asterisk shot the AST_STATE_PRERING correctly. But the DTMF tones are not reconized. In the chan_zap.c, the code: if (f->frametype == AST_FRAME_DTMF) { (...) Does not occurs because the frametype is always reconized as voice

Hi I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

I have an Asterisk 1.4 setup behind a TDM800P with 5 incoming lines, which happen to be inside my university's PBX. I have a problem wherein the university's PBX is apparently sending its CID spill after the second ring for a certain class of incoming calls (those from outside the PBX). I've temporarily hacked around this with a goto in chan_zap.c to restart CID processing if a good

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Morning, I'm thinking about buying the asterisk six-months online course, Have somebody here that bought that course? What is your opinion? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110606/3008703d/attachment.htm>

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

Anyone see this before? I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my colo. So as a did comes in they are routed to appropriate customers, in this case another asterisk 11.4 box. All is working well with the exception of DTMF. Losing the last digits so say someone hits 123... on the customer box I only get 12 This is the weird part, it only happens on 1 DID. If I

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 >>>

Curious if anyone else has run into this. I am testing this with a Sipura-2000 with firmware rev 1.0.33. On FC1 with Asterisk 1.0 Stable branch 4/1/2004 CVS (no April fools jokes please :) ) The Sipura has the ability so when you dial *67 it turns ON CID block and *68 turns it back off. (This is for outbound calls) When I *67 (activate CID Block) dial, and look at the SIP INVITE I see that it

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, I have connected a TDM400P to my asterisk, I have enabled DTMF CID, the data is arriving to the asterisk but asterisk isn't interpretating it: its my full log: 1. Mar 10 16:26:03] DEBUG[8715] dsp.c: dsp busy pattern set to 0,0 2. [Mar 10 16:26:03] VERBOSE[9274] logger.c: -- Starting simple switch on 'Zap/4-1' 3. [Mar 10 16:26:03] VERBOSE[9274]

Since the X100P/X101P/Clone cards does not work in all countries that use DTMF based Caller-ID systems, I've developed a hardware that you connect to a serial port and the PSTN. You then run a perl script "cid_logger.pl" as a daemon, and modify extensions.conf to call an agi script whenever a call comes in, and if it's on the X100 card it will get the caller id information

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