similar to: secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

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

Hello List, A few days ago I installed ViciDial on a server, and while looking to the default 'extensions.conf' file, I saw this line: exten => _010*010*010*015*.,1,Dial(${TRUNKTESTast}/${EXTEN:16},55,oT) Can someone point me out to the Asterisk documentation part where explains how to use server IP's as extension number? I could not see it in the ATFOT2 book, and I would

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

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

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

Hello Guys, After the Wiki was updated to the 3.5.X version, my username is no loger available: user: khratos mail: jpe at slackware-es.com I had some documents on my personal space. Is there a way to recover the account? Regards, -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs

Hello list, I'm facing a little issue with dahdi attempting to load the OSLEC echo canceller into my current kernel. After compiling dahdi 2.3.0.1 with OSLEC support, I get the following error when set 'oslec' as the echocanceller: DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) - Similar errors are *NOT* present using other echo canncelers. - I tried adding the

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi! Since some time the download of the newest Asterisk does not contains the version number anymore, but is just called "asterisk-1.4-current.tar.gz" This gives me a tarball where I do not know the version without looking into the tarball. Thus, IMO it would be very useful to switch back to old schema war the download contained the version number. Thanks Klaus

Hello Folks, What could be producing the following warnings on console, after an installation from source (Asterisk 1.4.41): [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module: Error loading module 'res_musiconhold.so': /usr/lib/asterisk/modules/res_musiconhold.so: undefined symbol: cap_set_proc [May 12 21:36:54] WARNING[15344]: loader.c:434 load_dynamic_module:

Hello List, Maybe I'm mistaken, but, shouldn't the 'fname_base' variable of 'Monitor' application affect the file name generated through 'automon' feature? I initialized this variable with a value as follows: Set(fname_base=auto-${CALLERID(num)}-${EXTEN}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) a. Should I use 'fname_base' in uppercase (FNAME_BASE)?

Hello List, Is it possible to select ODBC_STORAGE without entering to 'menuselect'? I'm currently building a package for my distro with a little script, and would like to set this option without entering manually to 'menuselect' I know that I could make the script to change the 'menuselect.makeopts' var from: MENUSELECT_OPTS_app_voicemail= to:

Hello, After installing Asterisk from source in Slackware 13.1, I get the following error: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache Then a core dump. If I change the /etc/asterisk/modules.conf in order to preload the 'res_odbc.so' module, then the error dissapears, *but* still crashes

Hello List, I have been working on a little PHP software that uses AMI's UpdateConfig command in order to modify some of it's config files. I was working with 'Asterisk 1.4.22.1' and everything was working. After upgrading to 'Asterisk 1.4.23.1' I receive a lot of errors of the type: ERROR[11505]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe

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

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

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