similar to: G723 format compilation errors

Greetings, I am trying to compile pam_winbind.so under Solaris 8. I have the source for samba 2.2.8 on the server. When I issue a make nsswitch/pam_winbind.so, I receive many compiler warnings, such as: Compiling nsswitch/pam_winbind.c with -fPIC nsswitch/pam_winbind.c: In function `converse': nsswitch/pam_winbind.c:72: warning: passing arg 3 of `pam_get_item' from incompa tible

Hi , I am facing a problem while configuring Samba on AIX 5.2. The configure completes without any issues , but getting an error while running make . Any pointers to this problem will be helpfull. Options used for Configure are - ./configure --with-winbind --with-ldap --with-ads --with-pam Here is the Error - # make Using FLAGS = -O -I/usr/local/include -I./popt -Iinclude

Hi , I am facing a problem while configuring Samba on AIX 5.2. The configure completes without any issues , but getting an error while running make . Any pointers to this problem will be helpfull. Options used for Configure are - ./configure --with-winbind --with-ldap --with-ads --with-pam Here is the Error - # make Using FLAGS = -O -I/usr/local/include -I./popt -Iinclude

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations

The following occurs during make asterisk-addons. I'm ok with asterisk but debugging things like this isnt my strong point. Can anyone give me a pointer? Thanks Dan Journo [root@sip1 src]# cd asterisk-addons [root@sip1 asterisk-addons]# make make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes

I upgraded * via svn and it did not work !!! 1. asterisk-addon did not compile! pbx:/usr/local/src/svn-versions/asterisk-addons # make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/local/src/svn-versions/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes

Anyone run a make on asterisk-addons /trunk r219 ? I error out on mp3 on a FC4 box, and I do not see anything obvious (to me) in the errors. make[1]: Entering directory `/usr/src/addons-trunk/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c gcc -pipe -fPIC -Wall

Ok, After following BJ's advice and removing ogg.so I then got a pbx_realtime.so error in the same fashion. I removed that file, and then the next and then the next as you can see in the log below. I think something is not right. duh here is my sign..lol...but I am not sure even where this ast_register_file_version flag is in a config file or what step I have missed. I am doing a VOIP only

Did the following: svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel svn checkout http://svn.digium.com/svn/libpri/trunk libpri Compiled and installed zaptel, libpri, asterisk and finally asterisk-addons. Following errors ocurrs when compiling

Hi all, I am trying to install samba on AIX. I have done a "configure", but while running "make" it gives the following error. Compiling nsswitch/pam_winbind.c with -O2 nsswitch/pam_winbind.c: In function `converse': nsswitch/pam_winbind.c:67: warning: passing arg 3 of `pam_get_item' from incompa nsswitch/pam_winbind.c:70: warning: passing arg 2 of pointer

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Good Morning, Any help would be grateful to help me understanding what's wrong... I have bought 2 g729a licenses to digium and I would like to have them works... My processor is an Intel(R) Xeon(R) CPU E5310 @ 1.60GHz (4 processors) so I have downloaded the

This *should* be something that's easy to fix, but apparently I'm not doing something right. Our SIP long distance provider is telling us to only use formats G.723 and G.729, so I've set up their trunk configuration in sip.conf as such: [t564] type=friend host=XXX.XX.56.4 context=default disallow=all allow=g723 allow=g729 However, the Dial application gives the following error:

Hi, I have licensed the digium G.729A codec. But for some reason incoming and outgoing calls will ALWAYS use G.711a. When I force my phone to only accept G.729 then an incoming call from ZAP goes straight to my voicemailbox as the phone doesn't accept the codec Asterisk wants, even if I force it in sip.conf. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The

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

Ok, I built a test system to duplicate my problem and provide myself a platform that I can mess around with to try and break any features. My problem is G729 pass-through from a gateway to a phone. I think I even have transcoding working, which makes me more confused on what's wrong with my pass-through. It must be a configuration issue. The basics... *CLI> core show version Asterisk

Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN -> Cisco -> Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call

Dear asterisk users, I am presently playing with an asterisk@home. I am trying to find the best codec solution for my voicemail records. I want to use ARI (Asterisk Recording Interface) to read the messages. I first used the default wav encoding that was not appropriate because my navigator does not handle wav mime types correctly and I have difficulties playing wav files in my basic linux

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

> > Maybe I'm missing something really obvious here, but I was unable to > > create a matrix out of POSIXct object(s). > > Perhaps that deserves a separate discussion...? > Can you provide an example? ------ #date and time objects x = Sys.Date () + 1:16 y = as.POSIXct (x) #matrices str (matrix (x, 4, 4) ) str (matrix (y, 4, 4) ) ------ Creating a matrix from a Date

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --