similar to: manager api events (pri vs pstn)

Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll

Hi all, I'm new in the list, and I have a problem upgrading from asterisk 1.2 to asterisk 1.4: There is a diference from asterisk1.2 to asterisk1.4 in AMI events. When I do a call to a queue (with the same extensions.conf dial plan) with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4 apper only 2. It is normal? anyone knows it? what is the reason? I

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on the analog cards does not work or i get cracking noise or even hangups on my BRI lines, due to bit slips.

hi all, i am trying to get my TE410P (see previous posts) working in Austria (telekom Austria - i am still waiting for an answer for my questions). my /etc/zaptel.conf looks like -------------------------------- span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 loadzone=at defaultzone=at -------------------------------- after

Hello, I get an Error every minute on the second card of two installed TE410P Cards in our System. The error is: PRI got event. HDLC Abort (6) on Primary D-channel of span 5(-8) PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 5(-8) Is it possible that there are known problems with 2 cards in one system? I'm running Asterisk/Libpri/zaptel from SVN branch-1.2-16008 I was

for today we only have experience with BRI applications together with asterisk. is the following scenario possible and stable enough for production? FYI : We want to build a unified messaging application integrated with SIP. We have an E1 connection in Belgium with 100 msn's We would think about having 2 servers : Server A : Asterisk PRI card (Digium TE410P) Server B : Fax

hello, i installed an asterisk as a pri gateway. Everything is okay. /etc/init.d/zaptel starts successfull with wct4xxp module. /etc/init.d/asterisk starts also successfully. I tested my pri cable and it works. But still my span isn't up. I don't see any error. Do you have any idea? What else i should check? Thanks. My card is 4 span Wildcard TE410P

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

Hi - We're moving up in the world to a PRI (Verizon), and I'm having some problems with it. I'm new to this PRI thing, though, so maybe I've just screwed up a simple config detail. I've got a TE410P on a Dell PE1600SC (ServerWorks Chipset). The card itself has a green light for the PRI, and Zttool shows Span1 as "OK". All I'm getting when I call into any

I'm trying to figure out how to start *..... Rh7.3,CVS,TE410P,TA750 If I just try the way the docs spell it out "/usr/sbin/asterisk -vvvc" it fails...... /var/log/asterisk/messages Oct 3 22:23:34 WARNING[1024]: File chan_zap.c, Line 610 (zt_open): Unable to open '/dev/zap/channel': No such device Oct 3 22:23:34 ERROR[1024]: File chan_zap.c, Line 4930 (mkintf): Unable

Those of you whom have a free Washington State phone number from ipkall.om will know that one has to use the number at least every 30 days or else the number becomes disconnected. I have 3 numbers pointed at my asterisk my which work very well but I still had the 30 day problem. Is there a way that I can program asterisk to make a call to my WA numbers so that they wont get disco'd? I'm

This may be better off on the developer list, but I thought I would see if I was way off-base before I went there. I am working on a manager CTI client (currently for windows but with hopes of porting it elsewhere later). [Hold/Reconnect] I have many of the features working. I can originate calls (using the call-in/call-out originate that the manager provides) display call-related information

Hi all! I'm trying to replace our old Siemens PBX with Asterisk. I've installed a TE410P card on a Linux box (Fedora Core 2 - 2.6.x kernel). I had no problem with compiling and installing modules and according to dmesg/zttool everything is ok. However, even though I've configured extensions.conf to play a greeting message when dialed, I can't get it to work. When I type

Any ideas on the following? (CVS 10/01/2003) Only reference I could find was a Zaptel change log update... 2003-09-02 18:23 martinp * wct4xxp.c (1.6): Get rid of the Double missed interrupt message every time you load the driver and an email refering this to serial console usage. Something I should worry about? Oct 3 22:48:01 cti-350 kernel: TE410P: Double/missed interrupt detected Oct 3

Hello all, I have a data frame dataa: newdate newstate newid newbalance newaccounts 1 31DEC2001 AR 1 1170 61 2 31DEC2001 VA 2 4565 54 3 31DEC2001 WA 3 2726 35 4 31DEC2001 AR 3 2700 35 The following gives me the balance of state AR:

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

hi, I'm trying to configure my newly acquired TE410P card to work as 4 E1 spans. This is supposed to be a drop-in replacement to the earlier E100P card. However, on loading the zaptel module it gets configured as T1 spans basically doing a 'cat' on /proc/zaptel/1 thru 4, it shows 24 channels per span. After this ztcfg fails saying 'ZT_CHANCONFIG failed for channel 97'.

Hi all, just for learning purposes i made a little gui frontend that visualizes incoming and outgoing calls in realtime, using the events of asterisk. I experienced a strange behaviour for outgoing calls. The callerid for the *called* person got changed to one of my own numbers, after the channels git linked. After looking into the flow of events i saw that asterisk keeps sending an

Hi to all. I have a problem in the AMI. Sometimes the AMI don't generate the event NewState when the exten of destiny is Ringing and sometimes don't show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,all Channel: SIP/17-00006fd6 ChannelState: 5 ChannelStateDesc: Ringing CallerIDNum: 4191920902 CallerIDName: 4191920902 Uniqueid:

Hi all OK this may sound like a good one but maybe someone can tell me. Simple context is - I want to unplug my existing conventional PBX from the Telco and place * with it's TE410P in between. Now the difficult part, the existing connection is E1 PRI (Q.931) with 6 B-channels. I need to be able to trigger a D-channel to the old PBX and a D-Channel to the Telco (Not BT!). Next I can put