similar to: Cant compile asterisk #error "You need newer libpri"

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 all, I need the zap channels going, but got the following error. What do I need to change in my configuration? Thnx. chan_zap.c: In function `zap_send_keypad_facility_exec': chan_zap.c:2309: warning: implicit declaration of function `pri_keypad_facility' chan_zap.c: In function `pri_dchannel': chan_zap.c:9292: structure has no member named `call' make[1]: *** [chan_zap.o]

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

I'm trying to clean up some notices/warnings that are repeatedly logged in *.Any Help would be appreciated as I'm not sure of the cause /solution. Here are the errors: Nov 17 15:53:38 WARNING[1217602880]: File chan_zap.c, Line 1321 (zt_call): cidspill already exists?? +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ /* Don't send audio while on hook, until the call

Here is a log from a recent call made out on a ZAP channel from a SIP phone inside my network. For some reason, CDR is billing time even though the "busy tone" was detected. It's also logging the call as ANSWERED. Is this normal behavior? Seems a little odd to me. I have this as the first 3 lines of my zapata.conf [channels] busydetect=1 busycount=3 CVS HEAD updated late

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

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

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Steve, 1) go to /etc/asterisk 2) open modules.conf for editing using vi 3) add this line: noload=pbx_wilcalu.so 4) Save the file 5) Restart asterisk Lightup the candles, open the Cabernet Savignon ( or whatever your prefernce) and call your girlfriend. ;) Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I am a newbie and want to step up to VoIP and switch from analog connetion to my Astrisk/Lineox box. Any suggestions on configuring Vontage and what to get/ask when signing up? > Has anyone experienced problems with Vontage and Asterisk. I'm using >Asterisk (Current Stable) and Sipura-841 phones. While talking on an >outbound call the transmission seems to fade out until the

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

hello , i try to install dahdi-linux/dahdi-linux-current.tar.gz ,dahdi-tools/dahdi-tools-current.tar.gz and libpri/libpri-1.4-current.tar.gz but got these error msg make -C drivers/dahdi/firmware firmware-loaders make[1]: Entering directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make[1]: Leaving directory `/usr/src/dahdi-linux-2.6.1/drivers/dahdi/firmware' make -C

Hi, Has someone successfully deflected calls using a Digium BRI board enabled asterisk ? How would you describe your experience ? Which Asterisk, Libpri and Dahdi versions are required for this ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111011/6e746640/attachment.htm>

Hi all, i do have the following setup ISDN BRI Line -> openVOX Card/Asterisk 1.6.2.6/libpri 1.4.11.2 -> Dialplan Dial DAHDI -> ISDN PBX -> ISDN Equipment The user on the ISDN Equipment das enable call forwarding - Teilrufumleitung / Call deflection - so that call will get forwarded by the telco switch - and not using b channels. The forwarding request is coming in on asterisk (i

Hi, I've recently met weird behaviour on 2 different and newly upgraded libpri1.4.12/2dahdi2.5 systems (at the moment, I can't correctly describe the symptoms but that's another story). For various reasons, this lead me to wonder which Dahdi/Libpri combination/version is the most widely used on this planet. Regards

Hi, I tried to install Libpri-1.4.3 after downloading from sites- www.asterisk.com and www.downloads.digium.com. But in both the case the problem is coming "AVC access denied". I am using Fedora core 8. I asked this problem earlier and got advice to disable SELinux. But many people adviced not to do this as it does not require and if it is demanding then there is a bug. I am very

The Asterisk development team has released new versions of three libraries used with Asterisk. They are: libpri-1.2.8: This release contains a number of bugfixes that had been unreleased for months, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily