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Hello!! I am new user to Wine. I have downloaded Wine-0.9.15 sources on solaris box and tried to compile a simple "Hello World" program using WineLib. I am getting following error: winegcc: -Wl,-G,-B,symbolic failed. *** Error code 2 make: Fatal error: Command failed for target `helloworld.so' Can anybody guide me as to what went wrong? I have following environment setup

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Hi!! I am not able to compile a test program on solaris using winegcc tool. Can anybody help? Here's the program source code: // file test.cpp #include <stdio.h> #include <windows.h> int main () { SYSTEMTIME lpSystemTime; GetSystemTime(&lpSystemTime); printf("Today is: %d/%d/%d\n", lpSystemTime.wYear,

We want to setup Samba as a domain controllser on our 3 sites , we'll be highly appreciated if you solve our queries : 1. Can Samba act as a backup domain server if incase Master Samba server is down then Backup Samba server can take place . Same like Windows concept if PDC is down then BDC will up automatically. 2. Can we configure SAMBA in that way that user can login from any location in

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

<div><font face=3D"Arial" size=3D"2">there is no audio. </font></div> <div><font face=3D"Arial" size=3D"2">I have made port fordwarding in the ro= uter 1. I=20 have opened ports 5060-5070 to SIP and 10000-20000 to RTP.</font></div> <div><font face=3D"Arial" size=3D"2">In the

Hello everyone, I am having a bit of problem getting MusicOnhold to play. I am running Asterisk 1.4 with MPG123 0.59 installed. And here's what i see in the debugging window of asterisk: -- Started music on hold, class 'default', on channel 'SIP/x123-082043d0' -- Stopped music on hold on SIP/x123-082043d0 Any idea why it is not playing the file at all? thanks

I have incoming analog and SIP DIDs that all ring multiple sip extensions with a Dial command as the first exten. I am curious to know if it's possible for the incoming caller to transfer out of the Dial command while in progress and dial a single extension? Thanks! jlc

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

I have a TE210P digium card that has 2 E1/T1 ports. the code in my extensions.conf file for span 1 is : [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 ; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This

Hey, all. I was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Hello may situation is the next: Asterisk <--> NAT1 (router)<---> internet <--> NAT2 (router) <--> x-lite ^ | ip phone (cisco) Asterisk and de cisco phone are in the same LAN. I want to make a call between the x-lite and the ip phone. I can do the call but there is only audio from de ip-phone

Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL:

it there a way to install software so that all users on my system can access it or am I stuck with installing the same software over and over for each user (huge waste of disk space )

Hi people!! i'm using samba with ADS authentication mode. when i try access the server samba, from windows, using hostname(\\serversamba) is showed a authentication screen, I enter my username and password, but isn't accepted. but when i try access via IP(\\192.168.1.1) all works. samba make the authentication with my username and password in AD(win2k box) . Why? sometimes this

Email eller vedhæftet fil afsendt fra din adresse (eller med din adresse som afsender) er blevet afvist fra Allerød Kommune. Spam og virus bliver typisk sendt under dække af andre afsendere og den blokerede email behøver derfor ikke oprinde direkte fra dig. (Husk dog altid at have et opdateret antivirusprogram på din computer.) Du kan evt. scanne din computer med det gratis' værktøj

Hi, I have a requirement, whenever a user comes into the conference, it has to announce the user name to all the person who are all available in the conference. I have used Meetme(,di) where i is to announce the user leave/join with review. I user used I also, which is to announce the user leave/join with out review. In both the above cases, it is prompting the user to say their

I want to use Asterisk as a general message delivery system here. That is, I want to be able to have a (shell, perl, etc.) script on my Asterisk server dial an extension, wait for it to be answered and then play a sound file and then hang up, or even wait for a response or reactions to some IVR. Certainly if I had a SIP library, I could have the script simply look like a SIP extension but that