search for: dail

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...

...k 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 for samba does not work with the dail up connection account. If I have not confused anyone, what should I do?

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 december 2017 13:44 > Aan: samba at lists.samba.org...

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

...y) exten => 123,103,Hangup() ;exten => 1,1,Playback(digits/1) # repeat vocaly the digit that pressed exten => i,1,Playback(pbx-invalid) exten => i,2,Goto(incoming,s,1) exten => t,1,Playback(vm-goodbye) exten => t,2,Hangup() [intrnal] include => outbound-local exten => 123,1,Dail(${DAVID},10) [outbound-local] gnorepat => 9 exten => _9X.,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _9X.,2,Congestion() exten => _9X.,123,Congestion() Thanks David

...t; wrote: > Hi sir > i read your response mail on asterisk user list i need help with > that dial plan which help if we transfer the call to any extension & it not > anser with 26 sec, then that transfer call come back to us . > so plz help me sir on thsi with that dail plan which you already > working out. > this is help as friend & if you need any help like this plz tell me i will > try my best. -- Atis Lezdins, IT Responsible of BEST Riga, atis at BEST.eu.org ICQ: 142239285 Skype: atis.lezdins Cell Phone: +371 28806004 [Tele2, Latvia] Work phone...

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 feature? Thanks, Chris

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 Dail in tab is shown in ADUC but when i try to open i get an e...

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 X...

...in(1111,ekKLmMNS,7010) where 1111 is conf number & 7010 is Admin user ityself Also for all other user i use like this with same conf number 1111 exten => 601,1,MeetMe(1111|Mps) when 2 user in conf & wait for Admin. then Admin Dail 600 but i found no response when admin dail 600 its blank no response at console. All Dail Plan config are ok. But for 600 its balnk. what i need to do so can Admin Handle user who dail 601 & enter in same conf room 1111.Means i want to controld all the users? Also What is the use of user para...

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

...modified version such as: plugin { quota = maildir:ignore=Trash } Which does not work. just in case it's a semantic problem, I also tried: plugin { quota = maildir:ignore=INBOX.Trash } which also doesn't work. I need other ideas of what to try to make it work if you please. -- Kenny Dail <kend at amigo.net>

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 m...

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...

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.