search for: dailed

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

I have a home network with my dial up connection on the windows 98 machine. Since I have installed SAMBA and configured the computers to see each other, I can no longer pull up a website on the windows 98 machine. The problem is that I have assigned a permanent IP address for the Intranet. I have to use DHCP for the dail up connection. Therefore the IP address that I assign to Windows machine

Readin : https://wiki.samba.org/index.php/Samba_AD_schema_extensions Is it an option to make an ldiff for the msNPAllowDialin and others on that Dail-in Tab. Im looking at the automount example. Hints tips? Greetz, Louis > -----Oorspronkelijk bericht----- > Van: samba [mailto:samba-bounces at lists.samba.org] Namens > L.P.H. van Belle via samba > Verzonden: donderdag 14

On Thu, 14 Dec 2017 12:23:43 +0100 "L.P.H. van Belle via samba" <samba at lists.samba.org> wrote: > Hai Rowland, > > > Even that msNPAllowDialin is a standard attribute, its not in my AD > anymore, at least not within the users fields. I think in time this > disapert wil fixing things.. This setup is running and upgraded as of > samba 4.1. but thank for

I think I have it configured properly. ztcfg -vv shows it as channel 1 and zttool shows it as OK. But I can't dial out. When I try, it shows it arrive in teh right stack, but then issues the following errors: channel.c:1676 ast_request: No channel type registered for '{PSTN-1}' app_dial.c:554 dial_exec: Unable to create channel of type '{PSTN-1}' = = Everyone is busy at

Hi, I have problems while trying to dial from simple analog phone that attached to my TDM400P card. No matter which number i press i immediately get a congestion tone. when calling from outside (e.g cellphone )to the line on port 4 and pressing extension #123 everything works fine and i manage to make a connection. I've plugged on port(Zap) 4 the analog line and on port 1 the phone.

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where to look in info, and if you don't have something working as expected - you're welcome

Hello, I'm trying to get the following situation. Someone calls an application on one of our asterisk server. In this application the caller will call a SIP client. (with the command Dial) The Sip client is connected with another asterisk server. (see below) Caller --> asterisk01 (incoming server) --> asterisk00 (outbound server) --> SIP client (X-lite) Do anybody now how

Hi, I have a question about the dial command. Is the following scenario possible. 1) - Our asterisk server had a successful outbound call. - Our asterisk server has to call another caller and when answered asterisk has to connect this call to the another outbound call. My first question is , do I have to this with a DIAL command, of a MEETME command? (A) -

Hi all, Is there a "speed dial" type application in *? The NEC PBX we currently use has a feature which allows any phone to access a system-wide speed dail database simply by keying the speed-dial number and pressing the 'redial' key from any extension. Even using a vinella phone on an sli the user can dial 77+speedial# and access this directory. Does * have a similar

Hi Thomas, Really Thanks for your advice. I have tried your suggestions but problems still remain. My findings is that some Windows, regardless of version, can map to Samba through dial-up networking but some just cannot. I try to find the difference between these Windows but failed. Regards, Neil Thomas Cameron wrote: > Neil - > > The problem you are having is possibly from one of

Hai,   Im reading : https://wiki.samba.org/index.php/VPN_Single_SignOn_with_Samba_AD    I wanted to use the "msNPAllowDialin" , in ADUC tab "Dail-in"  but i notices this one was gone/ i was missing this one : https://wiki.samba.org/images/8/88/MsNPAllowDialin.jpg  Admin pc, windows 7 64bit, samba 4.7.3.  AD Reinstalled it with the needed dll's from a win2008R2.   Now my

Hello list: I have dom0 that is supported PPP in kernel and i have installed ppp tools for dailing-up. I have a question , How should i use ppp dail-up in domU ??? First, I should recompile domU kernel thus support ppp ,right??? How''s next ???? Please help.... -- axa@HiNet <axanet@ms32.hinet.net> _______________________________________________ Xen-users mailing list

Hi All I need to set my Asterisk conference such way that , during confernce Admin Can kick 1 or all user , Same for mute fuction.As well as Admin can increase or decrease conf & user volume. for that i used MeetMeAdmin like this exten => 600,1,MeetMeAdmin(1111,ekKLmMNS,7010) where 1111 is conf number & 7010 is Admin user

Hi all, i am doing my master thesis on server perfromance evaluation i am using asterisk as sip proxy server and sipp tool as traffic generator... i have run basic testing of asterisk like as shown in website http://sipp.sourceforge.net/wiki/index.php/Howto_test_an_Asterisk_server_using_SIPp when i have copied sip.conf and extensions.conf from the site and run the client and server i

I can't seem to find the magic combination to get dovecot to ignore the trash folder when counting quota. I've got 1.0rc15 and using maildir quotas. Not being able to trash messages when over quota is quite annoying. the docs suggest the following: plugin { quota = maildir:storage=10240:ignore=Trash } but as my quotas are defined per user via mysql DB, I'm thinking that I need a

Hi All when I try to use auto-dial to connect to outside phone , my applications get executed before the caller attend the calls , this happens only when I call outside no , ie when I use Channel: ZAP/1/0507451111 in my sample.call file , if I use Channel:SIP/326 , it works fine my ?sample.call? file contains Channel: ZAP/1/0507451111 Callerid: Asterisk MaxRetries: 2 RetryTime: 10

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

Hi, I'm trying to get asterisk to auto-dail out. I created a *.call file with the the top of it being "Channel: Zap/1/2609944", which should have connected to Zap channel 1 and dial out to 2609944, but It did not do so, asterisk would say a call was completed to Zap/1/2609944 but I never heard that phone ring. So I tried just putting "Channel: Zap/1" at the top of

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.