Displaying 20 results from an estimated 500 matches similar to: "Witch kernel version may i use to run well asterisk"
2007 Feb 12
2
i m looking for a document that allow me to install well an asterisk server
re Hi,
I m looking for a good document that allow me to install zaptel libpri
& asterisk without errors, i ve a TDM400 & TE110P, so please can you
help me
Kind Regards
Younss AZZAYANI
KASTERISK.COM
2007 Feb 16
2
freepbx with ASTERISK 1.4
Hi everybody,
it's possible to configure freepbx 2.2 with asterisk 1.4?
Have a nice day
Younss AZ
KASTERISK.COM
2007 Feb 08
4
error when compiling zaptel-1.4
when i compile zaptel
make linux26
make install
i got these errors:
make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp'
make -C datamods clean
make[1]: Entering directory `/usr/src/zaptel-1.4/datamods'
make -C /lib/modules/2.4.27-3-386/build
SUBDIRS=/usr/src/zaptel-1.4/datamods clean
make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386'
make: *** arch/i386/boot: No
2007 Feb 13
1
asterisk-sounds doesn't exist in the sources, how can i get it?
Hello,
I can't find asterisk-sounds in the svn.digium server, i ve been got asterisk,
zaptel,libpri,asterisk-addons (1.2 stable version)
Thank You
Younss AZ
KASTERISK.COM
2007 Mar 15
1
Freepbx Incoming call's configuration
Hi every body,
I've set up a Trixbox Server with TE110P,all things seem to work
fine(Thank You Malling lists & irc & Forums), but i need your help,
i ve 30 numbre from 60 to 89, i need to specify for each sip extension
a Zap number
for example to call the sales service the caller must call 555-4570
and automaticly the caller will be redirected to the 202 ( sales
service ) so nobody
2007 Mar 17
2
SMS Integration and SMS commands
Hi i would like to preform the folowing integration
1 send SMS to my asterisk Systen, i hope to know a way to connect a simple
GSM or CDMA cell phone to asterisk (how can i connect a cell phone to
asterisk)
2 be able to put in the SMS string a command to generate a call (can i pass
thru instructions to my server via SMS ? )
3 after the the SMS is sent my Asterisk system will see the incoming
2007 Feb 14
1
CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20
hello my friends,
when i make a genzaptelconf i get this message
********************
CAS signalling on span 2 conflicts with HDLC with FCS check on channel
*******************
Any idea Please?
I m installing zaptel 1.4
i checked in "http://bugs.digium.com/view.php?id=7860" that it's a bug
but beacause i m a newbie in asterisk i can't undrestand what exactly mean
Thank You
2007 Mar 01
2
blieve i my TE110P or My teleco provider ??
hi eveybody,
after many test with your help and the irc channels help, i get the
led on TE110P green
with this config:
span=1,1,0,ccs,ami
=====> alarms OK Green Led
but the provider say that i have to set my span to this
span=1,1,0,ccs,hdb3,crc4
=====> alarms: YEL/RED
i can't make call's yet to test because they have to sync the
Modulator in the other side
so any remark?
is my
2007 Feb 13
7
error when compiling asterisk-1.4
hi,
when i type
********
asterisk-1.4# ./configure
******************
i got this error
****************
configure: error: C++ preprocessor "/lib/cpp" fails sanity check
See `config.log' for more details.
*****************
# vi config.log
***************************
...........;
cpp: installation problem, cannot exec 'cc1plus' : No such file or directory
..............
2007 Mar 04
1
Configurations Files of TE110P
please can someone send to me his files like zaptel & zapta if he si
using TE110P
thank you
2007 Feb 14
6
genzaptool from "xorcom"
hi every body;
i installed zaptel 1.4,libpri 1.4, asterisk 1.4, asterisk-addons 1.4 succefuly,
but i can't find the command "genzaptelconf, so i tink to install
"handy zaptel toolset"
please can someone tell me whitch package goes with zaptel 1.4,
i consult "http://updates.xorcom.com/rapid/pool/main/z/zaptel/" but ????????????
2007 Mar 16
2
SIP phone supporting more than 10 extension with a call transfer command
Hi every body,
can someone please tell me about a SIP phone that support more than 10
extension (free or not free ;) ) wich will be used in my company, i've
bought a SNOM but it just support 5 sip extension
Kind regards
2007 Feb 12
2
Problems Asterisk with Digium TDM400 card => he don't see the disconnect
Hi
i have a big problems with my asterisk .. i use a Digium TDM400P for
connect a
analog line.
And not all time (i don't know why) he don't see the end of the call and
anyone can call me
(occuped)
For that's work, i am disconnect the phone cable and it's good
anyone have a idea ?
bye
2007 Feb 26
3
Yellow or Red alarm on TE110P ????
i get this message with a red signal on TE110P card:
*****************
TE110P: span configured for...
Calling startuo (flug is 4099)
wcte1xxp: Setting yellow alarm
*****************
what does mean ?
thank you :)
2007 Mar 05
2
Rx+,Rx-,Tx+,Tx- of TE110P
Hi everybody,
i need someone to tell me the spins numbers of Rx+,Rx-,Tx+ and Tx- of TE110P
and also if you can tell me have to made a cable like that??
Modem Teleco <-----------Self Crosscable------------>Asterisk
Rx+ <--------------------------------------------------------------> Tx+
Rx- <--------------------------------------------------------------> Tx-
Tx+
2007 Feb 07
9
Digium TE110P
Helo,
I have problem with Digium TE110P connected to CISCO 3640 (port on
NM-HDV-2E1-60) wth PRI E1 link. I use CISCO now for testing but when I
tried with real PBX problem was exactly same.
I have this messages in Asterisk conole and log sometimes:
NOTICE[1115] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel
of span 1
Usually 2-5 such messages in series, can be repeated after 10
2007 May 15
0
PATH_MAX' undeclared here (not in a function) in asterisk!
hello, James FitzGibbon:
thank you for your help. i am very new to arm-linux and embedded linux. i think what you said is right. i am not very sure the steps i taken are correct. i post it here and please give me some help. it might be help other arm-linux users too. i installed all necessary libraries in my linux. if i just install asterisk under my linux. there is no problem. but when i
2006 Jun 14
3
SIP, Microsoft RTC, and Originate problem
Skipped content of type multipart/alternative-------------- next part --------------
Reliably Transmitting (no NAT) to 111.111.111.50:16666:
INVITE sip:111.111.111.50:16666 SIP/2.0
Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport
From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b
To: <sip:111.111.111.50:16666>
Contact:
2005 Jul 06
0
Need urgent help witch create mask rights
Hello, I've found this in the newsgroup and it works fine with the
inherit owner option.
The user is not able to delete the files he has created. Se below.
But my problem is that this doesn't work if you create a new folder and
the new create files in this new folder.
Any ideas how I can get the users to not delete the folder an the files
under the folder?
Thank's a lot.
On Tuesday
2007 May 08
1
Problems witch SPA3102.
Hello,
i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with
cdr. Well all I want is to receive incoming calls from pstn on specified sip
account (suppose 8000), and to initiate outgoing calls from all my asterisk
sip accounts through SPA3102 device. Someone can explain me what may i set
on SPA and asterisk to do this thing. Thank you for your support.
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