similar to: Understanding the right way to get started with multiple trunks/extensions

For some reason I didn't see David's reply by email, and have copy/pasted the following from the list archives to make my reply, sorry if that messes up anyone's threading. On 4 March 2015 at 12:15, David Duffett wrote: > If you would like to set things up via the GUI on your incredible PBX, > [...] I'm trying to avoid a GUI for now so that I learn something, but knowing

I have a data frame containing the results of time measurements taken from several cells. Each cell was measured in conditions A and B, and there are an arbitrary number of measurements in each condition. I am trying to calculate the difference of each measurement from the mean of a given cell in a given condition without relying on loops. >my.df id cond time 1

Hey all, I have a setup with two analog lines coming into and Asterisk 13 box with a TDM400P and it takes a lot of rings before asterisk takes over. I've traced this same box on two different analog providers so it probably isn't a problem with them. I DO have callerid enabled and not sure I can turn it off (if they will let me). Any other ideas of making it ring through faster? By the

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

Dear all, I'm trying to merge 2 dataframes, but I'm not being entirely successful and I can't understand why. Dataframe x1 State_prov Shape_name bob2009 bob 2010 bob2011 Nova Scotia Annapolis 0 0 1 Nova Scotia Antigonish 0 0 0 Nova Scotia Gly NA NA

Hi all, I have situation in that we are renting out a room in our office building to another company, including the provision of phone lines via our Aterisk box. To keep billing simple, we would like them to be billed separately for all their calls. We think we can achieve this by them leasing their own outbound IAX trunk (which our providers have said is no problem) and having all their

On 21/03/2010, at 10:38 PM, Skip Montanaro wrote: >> I don't know anything about Solaris, but your paste doesn't actually >> contain any errors, just warnings (unless I'm reading "ld: warning: >> relocation error:" wrong). It might help to run make without -j until >> it fails, and then use `make VERBOSE=1` to print the exact commands >>

On 20/06/2008, at 3:22 PM, Chris Lattner wrote: > > On Jun 16, 2008, at 2:12 PM, heguojin at ict.ac.cn wrote: > >> Has anyone succeded ? > > This sounds like the sparc backed doesn't have inline asm support, > which isn't a surprise. I'd suggest disabling bootstrapping. That was going to be my guess (PR1557 I think it is). If I get some spare cycles in the

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

On 09/02/2010, at 3:57 AM, Chris Lattner wrote: > On Feb 8, 2010, at 12:37 AM, Nathan Keynes wrote: >> Firstly, the BNE/BA pair should be reduced to a BE (I assume this is the responsibility of AnalyzeBranch and friends that you mention). > > Right. Implementing AnalyzeBranch will allow a bunch of block layout and branch optimizations to happen. > >> However I still

I've just started deploying these (well the T28P model) after years of Snom issues and they look pretty good (although the documentation is execrable; if you thought the Snom stuff was obtuse Yealink have got them knocked into a cocked hat!). Anyway, for provisioning I use HTTP with a DHCP entry like:- # # Yealink Phones # group { #

Hi, Currently if the linker encounters a pair of linkonce function definitions with different types, it raises an error of the form: llvm-ld: error: Cannot link file 'item.o.bc': Function '_ZN18st_select_lex_unit12first_selectEv' defined as both ' %struct.SELECT_LEX* (%struct.SELECT_LEX_UNIT*)' and ' %struct.SELECT_LEX* (%struct.SELECT_LEX_UNIT*)'

Quoth Steve Murphy... >Date: Mon, 12 Jan 2009 08:51:03 -0700 > >QUESTIONS: > >Which of the two would you see being useful to you? Obvious comment really but given LEG based CDR, one can determine the 'Simple' data but you can't work it the other way. I'd therefore find LEG based CDR more useful as the granularity (time on Hold, in Queue, Waiting on pre-xfer ring

Quoth: OCG Technical Support <support at ocg.ca> > >Although we've programmed the softkeys per the manuals, they seem to have no >effect (just dead). For example, our 57i is setup like this: I had similar problems and ended up using the speeddial inband functionality. FWIW, my 57i's setup like so: softkey4 type: speeddial softkey4 label: "*Park" softkey4

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

I have two Asterisk systems that can route to each other via a VPN with firewalls disabled for testing purposes. Each Server can see (tested via nmap) UDP port 5060 on the other. So... I thought that I could simply use a Dial command in Server A's config to place a SIP call to Server B... but it doesn't seem to work. Server A (192.168.1.33) has: exten => *136,1,Dial(SIP/90 at

Hi all, I've put together some preliminary patches to add frontend support for the sparcv9-* subtarget (ie 64-bit SPARC), modelled on the corresponding x86-64 code - do these look reasonable for inclusion? This doesn't address the codegen side of things yet (isel falls over when trying to actually emit 64-bit code), but at least bitcode generation looks correct now. Tested on

Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten => 200,1,Dial(Sip/112&SIP/113&SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the