similar to: Macros and Redirect with Manager API

Hello R-helpers, I'm trying to add labels to points in xyplot graphs, but I want lo label only those points which have a certain level of my grouping variable, and have encountered a few problems. An example dataframe that goes with the following code is at the end of this message. 1st step, adding labels (from another column in the dataframe) to my chosen points : I've figures this

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

Hi there, I've got a small problem with the zaphfc channel. No MSN of an any incoming call which comes trough the ISDN card (Acer ISDN, with HFC chipset and zaphfc driver) which will be forwarded to the SIP-Phone will be displayed. Always it will be shown "asterisk" an the Display. --- snip (zapata.conf) --- [channels] language=de switchtype = euroisdn signalling =

Hello, In setting up * for my company's office and remote employees, I have a question about how to log one username into * as either a SIP account, or IAX account. For example, we will be using SIP phones in the office locally to the * server, however some employees travel, and want to use IAX (as it's much friendlier with firewall/proxy setups than SIP) clients on their notebooks.

Hi all, I'm sure this has been done before but I just can't figure it out. On my * box I have a simple IVR: [tolc_menu] ; Welcome and information to callers exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,Background(welcome-to-tolc) ; Say Hello exten => s,n,Wait(1) exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter extension number if known, or

Alright, this may seem like something relatively easy to do but I must be missing something or had a neuron misfire. I am trying to get The Status lights on my Snom200 hardphones to display the status of each one of my PSTN lines in my Asterisk server. Current Config: 3 X100P cards Asterisk CVS-02/25/04-18:06:52 5 Snom200 phones I am currently using the following macro to dial out

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

I've been using the following macro to ring SIP and IAX devices for a few seconds, and then add on a cell phone if there is no answer on the SIP or IAX device. Periodic problems began a few versions ago and now the problem happens every time with 1.2.9 and 1.2.9.1. The problem is that when a call from the PRI falls through to voicemail, the call is dropped before the voicemail greeting

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

On Tue, Jan 19, 2016 at 01:46:05PM -0800, Sinclair Yeh wrote: > Updated VMWARE_BALLOON_CMD to use the common VMW_PORT macro. > Doing this rather than replacing all instances of VMWARE_BALLOON_CMD > to minimize code change. > > Signed-off-by: Sinclair Yeh <syeh at vmware.com> > Reviewed-by: Thomas Hellstrom <thellstrom at vmware.com> > Reviewed-by: Alok N Kataria

Hi, Does any one know when this series will be applied? Sinclair On Mon, Feb 08, 2016 at 11:41:57AM -0800, Greg KH wrote: > On Tue, Jan 19, 2016 at 01:46:05PM -0800, Sinclair Yeh wrote: > > Updated VMWARE_BALLOON_CMD to use the common VMW_PORT macro. > > Doing this rather than replacing all instances of VMWARE_BALLOON_CMD > > to minimize code change. > > > >

Hi, Does any one know when this series will be applied? Sinclair On Mon, Feb 08, 2016 at 11:41:57AM -0800, Greg KH wrote: > On Tue, Jan 19, 2016 at 01:46:05PM -0800, Sinclair Yeh wrote: > > Updated VMWARE_BALLOON_CMD to use the common VMW_PORT macro. > > Doing this rather than replacing all instances of VMWARE_BALLOON_CMD > > to minimize code change. > > > >

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Updated VMWARE_BALLOON_CMD to use the common VMW_PORT macro. Doing this rather than replacing all instances of VMWARE_BALLOON_CMD to minimize code change. Signed-off-by: Sinclair Yeh <syeh at vmware.com> Reviewed-by: Thomas Hellstrom <thellstrom at vmware.com> Reviewed-by: Alok N Kataria <akataria at vmware.com> Acked-by: Xavier Deguillard <xdeguillard at vmware.com> Cc:

Updated VMWARE_BALLOON_CMD to use the common VMW_PORT macro. Doing this rather than replacing all instances of VMWARE_BALLOON_CMD to minimize code change. Signed-off-by: Sinclair Yeh <syeh at vmware.com> Reviewed-by: Thomas Hellstrom <thellstrom at vmware.com> Reviewed-by: Alok N Kataria <akataria at vmware.com> Acked-by: Xavier Deguillard <xdeguillard at vmware.com> Cc:

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi, This problem may be related to a configuration problem but I believe it is a bug in the FOP since restarting the FOP server clears the problem. Here is the scenario: Using AgentCallBackLogin and have four agents logged in a call is made to one of the agents directly from an internal phone. Okay so far. Call is hung up and the same extension is used to call another agent okay again, no

I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area. This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works. [incoming] exten => 2019,1,Goto(test,s,1) [test] exten => s,1,answer exten => s,2,wait(5)

Replace all the open-coded macros for generating calls with a pair of more general macros (__PVOP_CALL/VCALL), and redefine all the PVOP_V?CALL[0-4] in terms of them. [ Andrew, Andi: this should slot in immediately after "Document asm-i386/paravirt.h" (paravirt_ops-document-asm-i386-paravirth.patch) ] Signed-off-by: Jeremy Fitzhardinge <jeremy@xensource.com> Cc: Ingo Molnar