similar to: 1.4 compile issue

hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' Generating input for menuselect ... menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... [CC]

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

I have a X100P and a TE110P in my Asterisk box. I can either get the X100P or the TE110P to work, but never both. Here's my zaptel.conf span=1,0,0,d4,ami e&m=1-24 fxsls=25 When I load wcte11xp and wcfxo, I will get this error. [root at asterix etc]# modprobe wcte11xp ZT_CHANCONFIG failed on channel 25: No such device or address (6) /lib/modules/2.4.20-8/misc/wcte11xp.o: post-install

Hi, Sorry for cross-posting but I need to tap into my most helpful mailinglists for this problem. I have had assertion failed errors from make before and never understood what they were or where they came from. But now that I have them consistently for ffmpeg and mplayer on everything smaller than FC2, which prevents me from updating both mplayer and ffmpeg (and I know lots of other

Hi Timo, I have another problem, this time with user shared folders: User "markus" shared the folder "ForTest" to test: SETACL "INBOX/ForTest" test akxeilrwts dovecot-acl and shared-mailboxes.db have been successfully updated. As user "test" the folder (and "#User" namespace) is not visible. When I configure the shared namespace with

My TE411p does not seem to like to have two PRIs from different telcos (span 1 and span 2). I can get one working, but not the other. However, if I use vpmsupport=0 when loading the wct4xxp module, they both work. But here is the problem, vpmsupport=0 disables the on board echo cancellation. Any ideas? BTW, here is zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs bchan=1-23 dchan=24

Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7

i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and share 1 VPN ( freeswan ). It would be nice if the 17 LTSP servers could use asterix on one

Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get two TDM400P cards in two machines attached to separate adsl connections (in two different physical locations). I'd then like to be able to plug POTS telephones into each of them (I understand I will need FXS

Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this?

Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone

Hi all, I am handed a project to setup *. The requirement is that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will

Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server & possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000

--- Raja Chidambaram <raja_chidambaram82@yahoo.com> wrote: > Date: Wed, 3 Aug 2005 06:27:05 -0700 (PDT) > From: Raja Chidambaram > <raja_chidambaram82@yahoo.com> > Subject: Phone interface hardware > To: Asterisk-Users@lists.digium.com > > Hello All, > > We are in the process of putting together an asterix > based phone network in our office. The