similar to: urgent kindly help Samba please ----------

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

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,

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

<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 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

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

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:

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

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

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:

Cool, I'll have a look at this too, thanks All! ___________________________________________________________________________________________ Mario Pio Russo, System Admin SWG IT Services Dublin, Phone & FAX: +353 1 815 2236, eMail: mariopiorusso at ie.ibm.com IBM Ireland Product Distribution Limited registered in Ireland with number 92815. Registered Office: IBM House, Shelbourne Road,

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

Thanks Sven, good Idea let's see if i am getting this right: 1) use MS ADSI editor to add few more attributes to the "users" class 2) use ldapmodify from my ubuntu server to populate those attributes would that work? ___________________________________________________________________________________________ Mario Pio Russo, System Admin SWG IT Services Dublin, Phone & FAX:

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 )

On 08/04/15 10:57, Mario Pio Russo wrote: > sorry guys , I think I didn't explain well. > > basically I have a samba 4 domain (created by upgrading+migrating a samba 3 > PDC). For every new user that we add to this domain I need to save some > additional info which are very specific for our company/department, but the > problem is that I cannot create custom attributes into

On 2015-04-08 11:57, Mario Pio Russo wrote: > sorry guys , I think I didn't explain well. > > basically I have a samba 4 domain (created by upgrading+migrating a samba 3 > PDC). For every new user that we add to this domain I need to save some > additional info which are very specific for our company/department, but the > problem is that I cannot create custom attributes