similar to: Finding difficulty in installing Asterisk

I install the Asterisk into a RedHat9, exactly like manual says, and I'm having the attached error message when try to install libpri. Please, help on it. -------------- next part -------------- [root@LinuxLab zaptel]# cd .. [root@LinuxLab src]# cd libpri/ [root@LinuxLab libpri]# make clean; make install Makefile:93: .depend: No such file or directory ./mkdep -Wall -Werror -Wstrict-prototypes

i have error in install module of tor2. but it compile good. what happen ? ivr2:/usr/src/zaptel # make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o

Hi there, I am attempting to set up a simple BRI and SIP based platform using * with the quadbri card (it's not sharing an IRQ). I enclose my zaptel and zapata.conf files. For the inital test I'm simply trying to connect to the * demo menu. The drivers compile (with a few warning that I believe aren't important - see attachments). chan_zap comiles with the warning: chan_zap.c:

Hi, My configuration is pri----*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. 1) inbound echo For outbound call(iaxclient-->pri), there is almost no echo. But for inbound(pri-->iaxclient), I can hear distinct echo. Can Sangoma a104 or digium te406p help this problem? 2)Today i received te406p. I know T1/E1 jumper. But how can i

I have the following setup: DS3 -> Cisco AS5400 -> H.323 (ooh323) -> Asterisk Inbound calls work great but outbound calls fail. So to check and make sure we have outbound calling ability on the DS3 we setup a Cisco Call Manager Express and it can make outbound calls both local and long distance with no problems. The failure code is Cause i = 0x8381 - Unallocated/unassigned number. We

This is a forward of a question from the SourceForge LLVM forum. If anyone has experience with Cygwin and building the C front-end there, please reply and CC Alex. --- Post by Alex Vinokur below --- =============== Windows 2000 GNU gcc 3.3.3 (cygwin special) =============== ./configure ./make I have got compilation error ------------------------------- make[1]: Entering directory

Hi all, I have installed Asterisk-addons-1.4.5. I was getting error cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory So, I did following steps: cp asterisk-ooh323c/.libs/libchan_h323.1.0.1 asterisk-ooh323c/.libs/libchan_h323.so.1.0.1 make install make samples It worked properly.But still I am not getting ooh323.conf in /etc/asterisk Please help me. Am I doing

Hello, I have a problem, and I'm looking for you help. When I dial certain number my calls are getting dropped. I initiate the call, I hear IVR, then I am being transfered to operator, and then suddenly I get ISDN DISCONNECT message. I had this type of problem some time ago, and I thought it was a problem on the other end. But now this is a second time it occurred and I want an expert to

> > >On Thu, 2003-11-13 at 16:50, Freddi Hansen wrote: > > >>>> >Freddi Hansen wrote: >>> >>> >>>>> >> with boards from Aculab, we are replacing Aculab boards with Digium >>>>> >> boards BUT we would need more >>>>> >> Digium boards IF we could use both Digium and Aculab cards in

Hi, I need help with the randomForest prediction. i run the folowing code: > iris.rf <- randomForest(Species ~ ., data=iris, > importance=TRUE,keep.forest=TRUE, proximity=TRUE) > pr<-predict(iris.rf,iris,predict.all=T) > iris.rf$votes[53,] setosa versicolor virginica 0.0000000 0.8074866 0.1925134 > table(pr$individual[53,])/500 versicolor virginica 0.928

Hi, I have installed GNU gatekeeper. Then I am trying to install asterisk addons. I gave make and then make clean. I worked properly. Then I gave make install. It gave following error. make[1]: Entering directory `/usr/src/asterisk/asterisk-addons/asterisk-ooh323c' cp .libs/libchan_h323.so.1.0.1 /usr/lib/asterisk/modules/chan_ooh323.so cp: cannot stat `.libs/libchan_h323.so.1.0.1': No

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

While testing a problem with "spontaeously" and "occasionally" rebooting asterisk, I came upon this problem: Program received signal SIGSEGV, Segmentation fault. [Switching to Thread -1210770512 (LWP 11346)] 0x002e3fe1 in pri_release_timeout (data=0x93a0478) at q931.c:2589 2589 q931.c: No such file or directory. in q931.c q931.c is in libpri, function

Hi Guys, I am calling out 416-999-1111 on Channel 1 of PRI and then calling 416-999-2222 on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/4169992222 -- Zap/2-1 is proceeding passing it

Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to "no" and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce <bruceb444 at gmail.com> wrote: > Futher check

I am trying to setup Asterisk to act like a PBX connected via a PRI gateway to a voice netowrk where Asterisk is doing outbound overlap dialing for calls that terminate via that PRI. AFter researching through the archives and online dcocs, I thought I had everyting setup right, dialplan configured for '_X!' and the 'overlapdial=yes' in the chan_dahdi.conf file, but when I try and

Dear; Thanks a lot for guiding me. Is it possible that the installation libpri-1.4.11.5 newer than the libpri-1.4.11.5-patch? Well, when I typed (note: I am trying to apply the libpri-1.4.11.5-patch for the libpri-1.4.11.5): libpri-1.4.11.5# patch -p0 -i libpri-1.4.11.5-patch It gave me that patched detected as shown below (example of one file, and I got same for other files): patching file

Hi, I tried to use Early Media: exten => 1,1,Playback(demo-thanks,noanswer) same => n,Hangup() But when calling my extension I do not hear the voicefile - I only hear the ring tone. In the Asterisk-Log I can see, that the voicefile is played. I got the same result when using "Progress()" in the first priority. I tried "pri set debug on span 1" and got the

Hi all. I am searching for a way to send SMS via our E1 PRI line. We are in Portugal and I have seen some Internet/TV/Phone providers (ZON for those who know it) who install normal phones with SMS support in landline. So I just found a page from PT (Portugal Telecom) stating that the SMC number is either 12999 or 129990 (

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.