similar to: Make error on ZT_EVENT_DTMFDIGIT

Hi, I've seen a few posts about this but no fix. Anyone able to help? Here's what I did: I configured a brand new machine with Redhat 9.0. I made sure that I had: bison cvs gcc kernel-source libtermcap-devel ncurses-devel newt-devel openssl1096b openssl-devel readline41 readline-devel zlib zlib-devel When I went to get Asterisk I did the following: cvs checkout zaptel libpri and

Hi; It is my first time installing an asterisk PBX system . I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard . Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

Dear List i have upgrade my * box with the latest CVS version of Asterisk Stable 1.0 and zaptel/libpri my system is MDK9.2 with 1 TE410P and seems work well for now but i have a little amount of traffic (25 IN/OUT calls) i only notice this Warning.. What kind of error is? ------------------------------- Apr 20 21:28:49 WARNING[147466]: chan_zap.c:5979 zt_pri_error: PRI: !! Got reject for

I used: cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc and in the same order I try to compile it. Asterisk ends with the lines below. It complains of a newer libpri, but I just did it a step before! What do I miss? chan_zap.c:62:2: #error "You need newer libpri" chan_zap.c:128: error: parse error before '<<' token chan_zap.c:133:1:

Hi, I am a beginner in asterisk. Implementing it in my dept in India using TDM400b card with asterisk, zaptel, libpri version latest of CVS HEAD Callerid on my system is coming tough. Asterisk doesnot finishes the callerid spill and Cancells it. After going through code in Callerid.c and chan_zap.c I found that my line is providing caller id of length 8867. Flow enters in zt_call and

I have a T1 PRI from the telco and have been trying to get it to work with the wildcard T400P. I know framing and coding is esf and b8zs. what else should I be putting into the zaptel.conf and zapata.conf. Dave Anderson

I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

I am getting the following error when I try to build asterisk. gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\"CVS-11/13/03-12:48:12\" -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\"

Hey all, I had a user complaining of calls which were dropping mid-conversation. I looked into the time of one of the calls, and saw the following: Apr 4 12:13:03 WARNING[6670] chan_zap.c: No D-channels available! Using Primary channel 28 as D-channel anyway! Apr 4 12:13:05 WARNING[6660] channel.c: Avoided initial deadlock for '0x82b8430', 10 retries! Apr 4 12:13:05 WARNING[6660]

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

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 all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

Hi! I am trying to install asterisk-0.5.0. For now I have installed libpri-0.4.0, now I'm getting the following error message when I do the make: chan_zap.c: In function `conf_add': chan_zap.c:836: `ZT_CONF_DIGITALMON' undeclared (first use in this function) chan_zap.c:836: (Each undeclared identifier is reported only once chan_zap.c:836: for each function it appears in.)

Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Calling from PSTN let extension 601 ring twice, hang up and starts over again to ring twice, ... If I pickup I do not hear on extension 601, and on the PSTN it is still signaling to ring. Can anybody enlighten me, please? extension.conf [incoming_88097074] exten => s,1,Wait(1) ;wait to get caller ID in. exten => s,2,Dial(SIP/102,20) exten => s,3,Voicemail(u102) exten =>

Hi All, I have installed chan_unicall and MFC/R2 successfully, and is runnign fine. But I noticed that once unicall is installed, asterisk CPU usage as reported by 'top', jumps to 99% every few seconds. I have no incoming calls, and I have even removed the E1 lines from card and I tried almost everything possible but I was not successful in determining the cause of this high cpu

Hi, Recently I did a fresh CVS checkout of asterisk and am getting the following errors on compile: chan_zap.c: In function `zt_train_ec': chan_zap.c:1078: `ZT_ECHOTRAIN' undeclared (first use in this function) chan_zap.c:1078: (Each undeclared identifier is reported only once chan_zap.c:1078: for each function it appears in.) make[1]: *** [chan_zap.o] Error 1 make[1]: Leaving directory