search for: partying

Hi Reid, > 2. Make sure you aren't trying to link in some old/crufty crtend.bc > file. Try doing "make uninstall ; make clean" in your crtend > directory if you think this might be the case. "make install" fails - there is no such target. > If those don't clear the problem, please tell me what version > of LLVM you're trying to compile

I''ve run into what I think is a browser bug related to using remote_forms pushed in an RJS update. In my controller, I have an action that creates a new model and returns a form for one of its children: def create_party party = Party.create render :update do |page| page.insert_html :top, ''party-list'', :partial => ''party_header'',

A stupid question but does anyone know how to express the actual colours used by the main Canadian political parties? I want to do a couple of ggplot2 plots and have lines or rectangles that accurately reflect the party colours. I can probably play around with RColorBrewer or something to figure it out but if some some already has got them it would save me some time especially with the NDP

Hi Chris, > Also note, LLVM 1.4 will be released in the next few days, so if waiting is > an option, you might choose to do so. Alright, I've got llvm and llvm-gcc from RELEASE_14 cvs and tried building it under cygwin. 1) The first problem is with llvm in SysUtils.c: int executeProgram(const char *filename, char *const argv[], char *const envp[]) { ................ execveTy

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Anyone tried the digium TC400B transcoding card? What are your opinions? Thnx

Hi, Feature originate can be used make call thro' the web. There is a parameter ,Async, in it. I set it to true but there is no effect. Actually, I want to do the following. What I know the function originate is: originate call ---> party A party A rings party A answers call party B rings, party A still hear ring party B answers and A & B connected. party A will feel weird when she

Hi We have come across a situation where we are loosing synch of party 1 & party 2 voice in call recording. Here is the scenario Party 1 initiate a call to Party 2 using AMI commands When both calls are connected, we bridge these 2 calls. Then we start recording of this bridged call using AMI Monitor command. Monitor command is invoked on Party 1 only. Then we put party 2 on hold. We

Hi all I have a model "party" that has a time field "starts_at". I have created a form to add new instances of this model. <%= select_date Date.today, :prefix => ''party'' %> Sadly I get the error: NoMethodError in Parties#add undefined method `month='' for #<Party:0x256a600> How can I fix this problem? I remember that I saw in

Has anyone noticed that Asterisk seems to always set the remote-party-id in a SIP invite to be the same value as the From: field? In most cases that isn't a problem. However, in the case of an attended transfer it IS a problem. The remote-party-id should be the party who initially called and the From: should be the party doing the attended transfer. This seems like a bug. - Doug.

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

Dear all, I have question but cannot explain without providing some context first: I want to calculate how many policy-connected coalitions between 7 parties are possible. I have positions on an one-dimensional scale for each party and I have sorted the parties on the positions (it is sorted from extreme left to extreme right, hence using a left-right scale). A policy-connected coalition

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Hi all, I know when doing a Dial, when the called party hangs up, we have a few different ways to redirect the calling party to other parts of the dialplan. In this case, I have someone who would like to do the opposite... When the calling party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

Hi All, We are experiencing a a problem when running calls over IAX with g.729. The call flow is as follows: Sip handset -(SIP)> Asterisk1 -(IAX)> Asterisk2 -(SIP)> Carrier The first Asterisk system is running 1.2 and the second is running 1.0. When using g726 from the handset all the way thru to Asterisk2(then 729 for the carrier leg) calls go thru fine, but when using g729, there

Hello everyone ! I have a link_to_remote which creates a Party from a ContactRequest. The action on the server creates the party, marks the contact request as processed, and then returns a redirect. According to my knowledge of HTTP, that is the correct thing to do. Unfortunately, Prototype is being too clever for me at this time... It follows the redirect, without notifying me. Anybody has

Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when

Hello List, I'm working on an autodialer project. At the moment I use the Originate application then I "throw" it to an extension where I Dial() the other party and then both legs are bridged. The problem is that the Dial() will only run after the Originate finish its bit and I have lots of wasted time or even worse, the remote party hanging the call because instead of a human

Thanks Richard. This is exactly the answer I was looking for. I'm now assuming that Asterisk 11 was using it's equivalent "bridge_simple" but I was getting confused because the only bridge module I saw in modules.conf was bridge_softmix. When I upgraded to Asterisk13 that would have been the only bridge getting loaded at first. Is it expected that if bridge_softmix handled a