search for: eslackware

...te is that I sent all the commands that my script executes on the AMI to a file (/tmp/debug.txt ) Then copying all the file content into AMI interface directly ( telnet [server_ip] 5038) and it executed all the commands without any problems. Thanks for your help, -- Jose P. Espinal http://eslackware.com

hello all, I have installed centos 5.5 ( linux text) and I have updated it with # yum install bison bison-devel================?ok # yum install ncurses ncurses-devel==========?ok # yum install zlib zlib-devel===============?ok # yum install openssl openssl-deve=======?ok # yum install gnutls-devel============ ==?ok # yum install gcc gcc-c++============?ok # yum install newt

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

...r that part. Note: Or might it be a fully dependent setting of ViciDial? In the installation documentation of VD says (just above the exten => ... line mentioned previously): ; local server extens: ; BE SURE TO CHANGE THIS LINE FOR YOUR IP ADDRESS! Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs

Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? Regards Bilal

...': /usr/lib/asterisk/modules/app_externalivr.so: undefined symbol: cap_set_proc [May 12 21:36:55] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'app_dahdiras.so': /usr/lib/asterisk/modules/app_dahdiras.so: undefined symbol: cap_set_proc Uname -a: Linux eslackware 2.6.33.4-smp #2 SMP Wed May 12 22:47:36 CDT 2010 i686 Intel(R) Core(TM)2 Duo CPU E4500 @ 2.20GHz GenuineIntel GNU/Linux gcc -v: Reading specs from /usr/lib/gcc/i486-slackware-linux/4.4.4/specs Target: i486-slackware-linux Configured with: ../gcc-4.4.4/configure --prefix=/usr --libdir=/usr/li...

...they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as well. I. The scripts are located at: http://packages.eslackware.com/slackbuilds/asterisk/ II. So far, they consist of the following: (Listed in recommended building order) a. newt-SlackBuild.tar.gz NewT Libraries b. libpri-SlackBuild.tar.gz LibPRI c. libss7-SlackBuild.tar.gz LibSS7 d. dahdi_linux-SlackBuild.tar.gz DAHDI Linux e. dahdi...

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing

...inside the Asterisk source directory) indicates that only this files mention that parameter: ./ChangeLog ./channels/chan_sip.c ./CHANGES: ./configs/sip.conf.sample Could someone please point me to documentation regarding this two parameters? Thanks in advice. -- Jose P. Espinal http://www.eslackware.com IRC: Khratos @ #asterisk / -doc / -bugs

Hello, I am looking for a dialer program, free or not, that allows me to perform scheduled calls, generate reports and let me upload sound files. Is there something that fits these features?. If there is not any product like I mentioned before I am interested to build this kind of software but I need ideas to make it useful for technical and non-technical people. I don't want to spend my

Hi all, I have a question regarding pending (zombie) SIP sessions: on Asterisk CLI, with command 'sip show channels' , I see two channels in use with callID and other infos detailed; also 'sip show inuse' give me same result (in terms of channels usage): Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message xx.xx.xx.79 209

Dear All, First, I am not so much experienced in Asterisk. I need asterisk to route the call to soft switch when the caller is not in its extensions list. And also when routing to soft switch, a number 4327 has to be added in the caller's number and then routed. I think its not so hard in asterisk. Please help me. Regards, Pratik -------------- next part -------------- An HTML attachment

...)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) a. Should I use 'fname_base' in uppercase (FNAME_BASE)? or... b. Is this variable independent of the 'automon' feature? Thanks in advice, PS. version: Asterisk 1.4.33.1 OS: Slackware Linux 13.0 -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs

...per connector to my DB system (MySQL in my case). I have verified that, after making the package the first time (with the script), the ODBC related modules are enabled. (cdr_odbc, func_odbc, func_realtime, pbx_realtime, res_config_odbc, res_odbc. Best regards, -- Jose P. Espinal http://www.eslackware.com IRC: Khratos @ #asterisk / -doc / -bugs

...al Info: Asterisk Verison : 1.4.39 Bash version : GNU bash, version 4.1.7(2)-release (i486-slackware-linux-gnu) OS : Slackware 13.1.0 PTLib : 2.8.3 H323Plus : 1.22.0 -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas

...: find . | xargs file | grep "executable" | grep ELF | cut -f 1 -d : | xargs strip --strip-unneeded 2> /dev/null find . | xargs file | grep "shared object" | grep ELF | cut -f 1 -d : | xargs strip --strip-unneeded 2> /dev/null Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs

...uctions on README file. These scripts might save you some times if you like to maintain your own packages and/or are planning to write an automated installation script based on customized packages. I hope this will be useful to some people around here. Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs

Hi I have followed https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS%2FRedHat%29, to my surprise there is only one config file by the name zapata.conf under /etc/asterisk/ There are no other config files. Any thing i am missing ? Please suggest/guide. Regards, Kaushal