similar to: Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite....

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

XLite does not support transfer... You have to buy their XPro -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite.... Hey Guys Im

Hey Guys Im trying to forward a call from the asterisk mainmenue to my second computer with X-Lite installed.. What I've done so far is this: Installed X-lite @my win PC.. X-Lite configuration: Menu | System Settings | SIP Proxy | default Display Name: mateo01 User Name & Authorization User: mateo01 Password: **** Domain/Realm: 192.168.1.** SIP Proxy: 192.168.1.** 192.168.1.**

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

Hey guys. Im trying to setup Music on Hold. If I transfer a call (with dial) I like to put the call on Music on hold.. Here's what I've tried so far: On my I extensions.conf exten =>1,1,WaitMusicOnHold(30) exten =>1,2,Dial(SIP/mateo,18) exten =>1,3,VoiceMail(1001) I have also added this line to [context].. So it looks like that: ;[context] musiconhold=default Additinaly,

I don't see how anything operating on the "result" of a return() call could be legal. The special semantics of the return() call is that it does **not** return control to the place it was called from, but rather to the location where its surrounding function(){} was called from. Mateo. -- Mateo Obreg?n. On Friday, 20 November 2020 22:52:58 GMT Duncan Murdoch wrote: > On

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Perhaps the parser should warn if you use return() at all. It is rarely needed and is akin to the evil 'GOTO' statement in that it makes the flow of control less obvious to the reader. -Bill On Fri, Nov 20, 2020 at 2:37 PM Mateo Obreg?n <obregonmateo at gmail.com> wrote: > I'm not thinking of complicated cases. > > This happened to me in a function that returns 10

On 20/11/2020 5:36 p.m., Mateo Obreg?n wrote: > I'm not thinking of complicated cases. > > This happened to me in a function that returns 10 minute slots > > slot <- function (seconds) { > return (seconds %/% 600) * 600 > } > > Obviously I found the issue while debugging and corrected my code with > surrounding parenthesis, but I was surprised that the

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hi everybody, I'm new to this matter and I spent three days in trying to connect one SIP Softphone to an Asterisk Box. I always get error 401 or 403... I don't understand very well settings in Softphone program: con anybody show me how to set up a minimal running system with no public lines or external Proxies? Thank you for your kind help... Ciao Mauro

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

Hi all, I can confirm this occurs for me as well. The one thing that comes to mind is that there are certain larger expressions that contain calls to return which we absolutely don't want to be an error, e.g if(somestuff) return(TRUE) That said, the actual expression Mateo pointed out certainly does look like an error (it definitely isn't going to do what the developer intended).

I'm not thinking of complicated cases. This happened to me in a function that returns 10 minute slots slot <- function (seconds) { return (seconds %/% 600) * 600 } Obviously I found the issue while debugging and corrected my code with surrounding parenthesis, but I was surprised that the R parser did not catch this syntactic error. This is especially poignant when we have to

FWIW, 'R CMD check --as-cran' in R-devel checks for "bogus return" statements but I think that's only for the case when one forgets the parentheses, e.g. 'return' instead of 'return()'. I don't think it catches this case but I'm also not sure. Though, I can imagine it might be possible to enhance the current check to include also this case. It could

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

And the related: > f = function() stop(return("lol")) > f() [1] "lol" I have a feeling all of this is just return() performing correctly though. If there are already R CMD CHECK checks for this kind of thing (I wasnt sure but I'm hearing from others there may be/are) that may be (and/or may need to be) sufficient. ~G On Fri, Nov 20, 2020 at 3:27 PM D?nes T?th

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome!