similar to: Is Asterisk a PBX?

Hi, I'm about to incorporate scp2 support to our current SSH2 server implentation and I could't find any document (RFC, internet draft, etc) about the implementation scp or scp2. Does anyone know where I could find some ??? Thanks -- Charles Gelinas Software Engineering Ericsson Inc. Datacom Networking and IP Services Access Product Unit charles.gelinas at ericsson.com Ph: (805)

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call

While editing a record I need to have a link to create a new record in another scaffold. I''ve done this by using a popup window <%= link_to("Add new Person", "/person/new", :popup => [''window_name'', ''width=600,height=600'']) %> It all works great - but when the save is done I want to kill the popup window.

Hello All, Figured I'd ask this. I have an asterisk running with some X100P clones. The system runs fine. My question is, the lines are tied into a panasonic dbs analog extension. When you call through, all goes well except that the call does not know it is connected. This causes problems if lets say, I want to do a 3 way conference, since my cisco 7960 does not see the call as connected,

Hi all, What are the best and free tools for remotely adding, removing users on Asterisk server and also for monitoring the status of the Asterisk server, like how many users are logged on etc. I need tools for which I don't have to pay. Thanks, Zeeshan A Zakaria www.acabling.com <http://www.acabling.com/> -------------- next part -------------- An HTML attachment was

I'll preface this by saying I'm a sales person and am not super technical. Question: We are looking to implement an iSCSI SAN that will be accessed by Windows and Linux servers. We want everyone to have access to all files so the question is: will Samba on the Linux servers allow them to access the same files the Windows folks can? All servers have iSCSI initiators on them. If you

Hi, though the list would be interested to know an issue we have just come across. We have found that rsync version 2.5.6 over ssh within a cygwin environment run manually from either a command prompt or a cygwin prompt hangs every time if we run it in a Windows Terminal Server session. It doesn't seem to matter what the remote server is as we have experienced this when the remote server is

I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Hi, I've read through the Asterisk handbook and I'd just like clarification from someone that's implemented the above before. Lets imagine I want to use the CallingCard application and don't want to tell a client to buy a channelbank (the analog extensions are too many to connect to FXS cards), I figure I could set them up as below: Original Existing Setup

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

James Peltier writes: > Welcome to the world of UEFI. Certain UEFI versions have added additional support for things like the next-server option to actually be honoured. In some versions this _is_ in fact _ignored_ and you are expected to place the image on the server that answers the DHCP response. > > I have been successfully booting both legacy BIOS and UEFI machines for at least 3

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi, folks. I'm senior online editor at CIO.com, and I'm working on an article for which I'd very much like your help. There's often a lack of communication between techies and top company management. Maybe they don't want to hear about problems; perhaps you give them technical details that are far more granular than they want to know. But dealing with spam is a topic

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please