similar to: PLEASE HELP!! CALLERID FAILS!!

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

The 1.2 beta1 works fine. When I install the current cvshead it gives me different errors: I have seen checksum errors, Got event ring 18, etc. all give empty callerid. I have an x100p. Thanks --john

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

Hi all, I just installed an TDM02B. My system is a dell pc with linux 2.6.12-1.1456_FC4 asterisk-1.2.0-beta1 zaptel-1.2.0-beta1 libpri-1.2.0-beta1 in /etc/zaptel.conf I have (all others are default): fxsks=3-4 <--- I saw light in the ports channels=1-2 <--- change it to 3-4 has same result but... [root@nmsd0 asterisk]# /etc/rc.d/init.d/zaptel

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

I've got a rather strange callerid problem and don't know how to debug it. I've got two X100P cards installed in a RH9 box and both are connected to individual pstn pots lines. Both are equipped with callerid and both display callerid info on regular analog phones (verified 100%). Incoming and outgoing calls are handled just fine. When an incoming call arrives on one X100P card, *

Hi list! Do anybody has success histories about using the Beronet app_saynumber application? For those that are hearing about for the first time, it?s a piece of software that enables the use of another language in say_number commands in asterisk dialplan or AGI scripts. Link to download: http://www.beronet.com/download/app_saynumber-beta-rc1.tar.gz I?m trying to compile it in

I posted this 1.2.0-beta1 success story to asterisk-dev, and someone recommended that asterisk-users might benefit from it as well. Thanks, Alan Ferrency pair Networks, Inc. alan@pair.com ---------- Forwarded message ---------- Date: Thu, 22 Sep 2005 17:35:08 -0400 (EDT) Subject: [Asterisk-Dev] Re: Ring requested on channel already in use To: asterisk-dev@lists.digium.com > alan wrote:

The Asterisk development team is very pleased to announce that we have released the first 1.4 beta packages of all four of our projects! The beta versions are: Asterisk - 1.4.0-beta2 (beta1 was not released to the public) Asterisk-Addons - 1.4.0-beta1 Zaptel - 1.4.0-beta1 libpri - 1.4.0-beta1 All of these releases include substantial new functionality and performance improvements. The

The Asterisk development team is very pleased to announce that we have released the first 1.4 beta packages of all four of our projects! The beta versions are: Asterisk - 1.4.0-beta2 (beta1 was not released to the public) Asterisk-Addons - 1.4.0-beta1 Zaptel - 1.4.0-beta1 libpri - 1.4.0-beta1 All of these releases include substantial new functionality and performance improvements. The

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Asterisk version: 1.2.0-beta1 -OR- CVS HEAD Hardware: Generic X100P clone connected to Panasonic KX-TA624-5 extension port The problem happens IF AND ONLY IF: - the Panasonic is set to use double rings - the X100P is set to answer immediately (I'm using DISA here) It does not happen consistently; sometimes Asterisk behaves normally. When the "strange state" happens, DISA usually

Hi, I was about to send a question to the list about this when I figured out the solution. I thought it might be useful to store in the archives. I use a X100P and a TDM400P, using current code from CVS. When I dial in via the X100 I get -- Starting simple switch on 'Zap/1-1' right away in the CLI. However, it takes (took) * a few seconds to continue patching the call through to

The first beta of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta1' tag). This version of Asterisk represents a significant improvement in features, stability and compatibility over the 1.0.x releases. Some of the major new (or upgraded) features include: * Asterisk Realtime Architecture

Hi all, With a great help from Richard Alexander (thanks Richard!) I have now a functional CallerID on my X100P. This is what I have done: - update to the latest CVS (as today at 5:00pm GMT) - modify the callerid.c file in the asterisk source like that. original : /* MDMF */ /* Go through each element and process */

I'm getting this error on incoming calls on my X100P cards: -- Starting simple switch on 'Zap/1-1' Oct 7 15:49:19 ERROR[74769]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-2) Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5369 ss_thread: CallerID feed failed: Success Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5411 ss_thread: CallerID returned with error on channel

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

Hi all, In order to get the CallerID from PSTN (X100P) I have modified callerid.c file like that: callerid.c [line 256] from: case 3: /* Number (for Zebble) */ to /*case 3: Number (for Zebble) */ Without this modification my own number was displayed as the inoming call CallerID. Now I want to go further. I have activated CallWaiting support on the POTS line. When someone calls and I am in

I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic i want to configure a x100p card an use it with asterisk, so i download, compile and install: asterisk-1.4.24 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 i try almost everything i found on the net but without success: if i run lspci: 04:06.0 Communication controller: Motorola Wildcard X100P when i run dahdi_hardware appears this:

>1- Call comes in without callerid >2- AGI script answers line >3- AGI script asks to record name >4- Park the call and get the parked extension number >5- Ring all the phones in the house (exec Dial) >6- If phone is picked up, play recorded name >7- Wait for DTMF to accept or decline call >8- If accepted, bridge parked call and current call. Mike, I am wanting this