similar to: Dialer program

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

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 have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

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 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 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

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

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

I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is

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

One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/325c0670/attachment.htm

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

I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a

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

Hi Tilghman!   > Clearly, you missed the point.  Since there is a FREE predictive dialer out > there, and your product costs something, you are not the world's cheapest > predictive dialer.  I respect your wording and the way you or other people think on the list about difference between cheapest and free predictive dialer.   Surely