similar to: TDM - Analog Trunk - CallerID question

Dear list, I am trying to learn how to use Zap-things in Asterisk. While loading Asterisk verbosely I get this error: [chan_zap.so]Warning, flexibel rate not heavily tested! => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Mar 12 17:19:01 WARNING[5563]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 12

Of course I am not a kernel expert, so .. please be patient. I am investigating on my zaptel/zapata problem. As the main error message asterisk quits on mentions <'/dev/zap/channel': No such file or directory> I went peeking over there. [Asterisk Verbose Error Mar 13 20:43:35 WARNING[5779]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or

I know that this subject has been treated in the past! As a matter of fact reading some old messages about compiling zaptel I made a couple of tests after the first compiling failure to understand why I can't compile on a specific machine, but I do not know how to handle the results. The machine has SUSE 9.3, and an updated kernel (2.6.11.4-21.7-default; as shown below). YAST (the graphical

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi list, I am still attempting to start an asterisk 1.0.6 fresh installation. There are some problems with the zap channel: == Parsing '/etc/asterisk/zapata.conf': Found Mar 13 10:49:38 WARNING[5278]: chan_zap.c:763 zt_open: Unable to open '/ dev/zap/channel': No such file or directory Mar 13 10:49:38 ERROR[5278]: chan_zap.c:6208 mkintf: Unable to open channel 4: No such file

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

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

Hi all, I've been trying to wrap my mind around this one for several days now. How can I 'debug' the CallerID reception on a Zap/POTS channel? I have a POTS line with CallerID and a Digium TDM11B card right now. I have my signalling set to ks for both sides, can make and receive calls just fine. But I never get CallerID on incoming calls. I get the following messages: Aug 11

A very good day to you all, We can't get the phones to pick up on an incoming call on analog trunks. We're using the digium products in the box, with snom phones internally. This is the output from the asterisk console: linux*CLI> zap show channels Chan Extension Context Language MusicOnHold pseudo pstn-incoming en default 1 pstn-incoming

Yo! I changed callprogress to no, and in wcfxo.c source around line 334 i changed the value 32000 and -32000 to 10000 and -10000 because it had something to do with the DC voltage when it was ringing. I found reference here (http://www.voipuser.org/forum_topic_1791.html) with an interesting diagram of wiring that was incorrect for sending voltage to a phone or something like that. So put it

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

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got my developers kit from telappliant and got a machine up and running to become the house phone system. Most things are working now, such as incoming calls, call transfer, call parking, voicemail, etc. The one thing I can't do is make my analog phone ring! I can see the call coming in on the asterisk console and can then pick

Hi There, We have a Asterisk 1.4 box with a X100P card connected to a analog line with Caller ID serrvices enabled on it. When an incoming call appears we get the following in the log: -- Starting simple switch on 'Zap/1-1' -- Detecting post-CID distinctive ring [Apr 15 10:38:07] NOTICE[7151]: chan_zap.c:6469 ss_thread: Got event 18 (Ring Begin)... [Apr 15 10:38:07] NOTICE[7151]:

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

HI, im having problem with callerid. Im using tdm2400P and i get this from asterisk logs -- Starting simple switch on 'Zap/4-1' [Mar 24 02:07:48] ERROR[2358]: callerid.c:564 callerid_feed: fsk_serie made mylen < 0 (-1) [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6416 ss_thread: CallerID feed failed: Success [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6516 ss_thread: CallerID

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

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

As everybody must be watching the superbowl. I post this to let you have some fun while thinking what this can be. TDM400p (fxo) connected via loopstart to ports in an AvayaG3 call comes in from the avaya to the tdm card: WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' but call can be processed normally. comments? --

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 : --