similar to: Documentation

Hi there! Has anyone succesfully used a Cisco 12 SP+ with *? If so, how did you do? I'v not even tried, but before trying I thought I could bug you somewhat. =) //Filip

Hi All, When the Caller hangup at the voice menu, the wildcard X100P didn't disconnect the calls properly and it just keep looping at the voice menu and timeout and loop again, are there any methods can fix the problems? Please help! Thanks, Randal

I''m hoping there''s an easy way to do this.  I''m trying to read options from a database, and insert them into form tags.  So for example, In the Database Table: options size: 300 type: text_field maxsize: 300 and I want to create a text_field form tag, with all the appropriate options, is there an easy way to do this, other than having the code output the HTML

I had fun with that - enter any number you like, if it's not valid it will prompt you with suggestions. It seems to want decimal on the commandline, hex if responding to the prompt. Phil Phil Randal Infrastructure Engineer Hoople Ltd | Plough Lane | Hereford HR4 0LE Tel: 01432 260415 | Website: www.hoopleltd.co.uk Follow us on: Twitter | Facebook | LinkedIn -----Original Message-----

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,

What does vga=819 and vga=319 mean? On Wed, 25 Sep 2019 at 01:54, Jon Pruente <jpruente at riskanalytics.com> wrote: > > On Tue, Sep 24, 2019 at 6:53 AM Randal, Phil <phil.randal at hoopleltd.co.uk> > wrote: > > > Workaround is to select the install option, hit the tab key, and append > > vga=819 and hit F10, which starts the graphical installer on a larger

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

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

Hey all, Ok this is probably a newb question, but I haven''t been programming with MVC for very long, and I''ve been wondering what the general opinion is, or best practices, for making a new controller. Like what cases would you need a new controller for and what cases can you just put everything into one controller? Thanks in advance, Randal -- Posted via

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to

http://voip-info.org/wiki/view/Easy+PABX With Easy PABX you can create your own virtual PABX online in just minutes. Easy PABX is based on Asterisk and best of all - it's completely free. Regards thorben.dk

I am installing one asterisk, to establish connection with my PABX Siemens, in ISDN, link went up normally, also I obtain to internally call the branches the PABX, normally, but when I try to dial for the PSTN, through pabx with the command exten = _ 19xxxxxxxx, 1, dial(zap/g2/${EXTEN}, 30) asterisk, reports me the following error: -- Executing Dial("SIP/8110-a729",

Noob Question 31,265,232 if I''m searching on an object, say order, that has many "order_lines" and I want to display order lines by Quantity ( an attribute of the order_lines ) descending how could I do that without having to do a find() with :order, but something like; Order.order_lines.each do |ol| <!-- code to diplay the line --> how can I determine the field

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 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 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

How do I get * to take an incoming oh323 call and let it dial a number? I.E. if my boss sets up netmeeting with the gateway as my.pabxbox.com, whenever he enters a number to dial it always just dials into the pabx rather than calling that number i.e. he wants to call 12345 he types it in and presses dial but it just goes to the message.... also I have developed an H323 client app which I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello I am running an * box with just 8 extensions connected to our old Alcatel BCN 5200 PABX. The requirement is that we now scale it up to handle about 300 lines and get rid of our old PABX. Is there a way of hooking up 300 phones to asterisk without going via the PABX. I am more of a network person than a telecomms one so i may not fully

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT