similar to: asterisk-addon

Hello maybe someone can help me? I did the CVS checkout and then compiled asterisk Then I tried to compile the addons and got the following (don't understand what's wrong at all and can't find anything about this error on google/wiki) app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function

Hello I have downloaded asterisk-addons but when I make install get: cc -fPIC -I../asterisk -D_GNU_SOURCE -DMYSQL_LOGUNIQUEID -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error:

Hello, i have a little Problem with compiling asterisk-addons the failure is: app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function `del_identifier': app_addon_sql_mysql.c:164: error: ?SR_LIST_REMOVE' undecalred (first use in this function) app_addon_sql_mysql.c:164: error: (Each undeclared identifier is

Hello I am tying to install asterisk addson, but when i give make, it shows me following error messages: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:164:77: macro "AST_LIST_REMOVE" passed 4 arguments, but takes just 3 app_addon_sql_mysql.c: In function `del_identifier':

good day all, my asterisk is working greatly.. and i want to put a billing.. but i have this error when i try 'make' [root@upm asterisk-addons-1.0.7]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -I/root/asterisk/include/ -c -o app_addon_sql_mysql.o app_addon_sql_mysql.c app_addon_sql_mysql.c:165:77: macro "AST_LIST_REMOVE" passed 4 arguments, but takes

Hello everyone, Version 1.0.7 of Asterisk, Zaptel, libpri, and Asterisk-addons has now been released. Libpri and -addons have not changed, but have been updated anyway to keep the version numbers consistent. All of the tarballs are available on the ftp site. ftp://ftp.asterisk.org/pub/asterisk/ I have posted the ChangeLogs for easy viewing at the following address.

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

there is a problem in compiling asterisk-addons any one have fixed this problem. i want res_config_mysql.so any one help me ----------------------------------------------------- [root@localhost asterisk-addons]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c: In function `realtime_mysql':

Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus

Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to someuser@myname.co.za it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the

Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added "exten => 403,hint,SIP/403" in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? Please Help Altus

Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

Good day all I got my voicemail message working,thanks but now,keep in mind I'm using SIP We have,for example 4 people in our admin department.Each user has its own voicemail so that when their extension is dialed directly and not answered it gos to voicemail. But there is also a option to dial 3 for admin with will dial all 4 number in sequence.This I got working 100% but now I want a

PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has

Hi there: I have successfully installed the Asterisk 1.0.9 on my Freebsd 5.4 box. When I tend to install the addon for mysql CDR billing, It always return me the following errors: SIP# gmake clean rm -f *.so *.o .depend gmake -C format_mp3 clean gmake[1]: Entering directory `/usr/home/wilson/asterisk/asterisk-addons-1.0.9/format_mp3' rm -f *.o *.so *~ gmake[1]: Leaving directory

Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he call....BUT there is no sound.It shows there is a call and you are