similar to: Asterisk Legacy PBX

Hello Everybody I am using Trixbox 2.4 (with TE420P & PRI lines) .. my setup is like Calls -->Asterisk-->legacy pbx--->analog extensions(agents). Whenver a call comes in , asterisk dials the ACD number of the legacy pbx which in turn decides to route to appropriate agent.. for ex : s,1,Dial(ZAP/g4/5432) [g4 is the 4th span and 5432 is the ACD number of legacy pbx under which agents

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

I have a site running 1.4.17 with Zaptel. They want to add a TE420P for additional T1 capacity. I'm 99% sure this will work, anyone aware of a reason it wont? Thanks, James -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100806/b5f4bc0f/attachment.htm

Hello I am trying to configure TE420P but i am confused what to give chan_dahdi :( Below is configuration i am using for TDM400P Please help what changes to make in it... Please provide a link as well [trunkgroups] [channels] ;default for channels switchtype=national rxwink=300 hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

I'm putting together a PBX using a TE420P card configured for E1s that is connected to an Errickson MTS. successfully compiled and installed libpri 1.4.11.3, DAHDI 2.3.0.1+2.3.0 and Asterisk 1.6.2.9. everythings seems to be working. SIP phone to SIP phone (POLYCOM) calls work fine however, network calls do not. When I went to debug PRI, the only command that showed up when I did ">CLI

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Hello I have recently configured TDM400P with four FXO ports. My next requirement is to configure for E1 line. which contain 30 phone lines and 2 for signalling information. The problem is I dont want to go for E1 line directly .....Is it possible to get simulation for E1 line ... so that i can develop a system for an E1 line. -- Best Regards Shakeel Abbas

Hello, This patch aims to improve compile time performance by increasing the SCEV vector size in LoopStrengthReduce. It is observed that the BaseRegs vector size is 4 in most cases, and elements are frequently copied when it is initialized as SmallVector<const SCEV *, 2> BaseRegs. Our benchmark results show that the compilation time performance improved by ~0.5%. Patch by Wan Xiaofei.

Hi Daniel, I made some changes to the LNT perf reporting tool to make it more user friendly by adding some features: 1. Make the sidebar and the navigation bar stationary, so that it is easy to navigate the site 2. Have the pop-down menu for the items in the navigation bar, activate upon hovering the mouse, rather than clicking the item 3. Add a nav-link in the sidebar for the

Hi, I am trying to get a small loop to *not vectorize* for cases where it doesn't make sense. For instance, this loop: void foo(int a[4][8], int n) { int b[4][8]; for(int i = 0; i < 4; i++) { for(int j = 0; j < n; j++) { a[i][j] = b[i][j]; } } } * Has maximum of 8ints copy. LLVM tries to use Memcpy for the inner loop. It is not helpful to perform

Hello there, I recently started working on the LLVM backend for a target that doesn't support x87 instructions. Currently, I am in the process of completely disabling some x87 instructions such as fcomi, fcompi,... for a specific sub-target. I also do not have SSE enabled for my sub-target, and llvm resorts to fcomi* instructions for FP compare instructions. Is there a way to bypass the

Hi Nadav, Thanks for the response. I forgot to mention that there is an upper limit of 16 for the Trip Count check, TinyTripCountVectorThreshold = 16; if (TC > 0u && TC < TinyTripCountVectorThreshold). So right now, any loop with Trip Count as 0, or with value >=16, LV with unroll. With the change to the lower bound, it will also include the loop with 0 trip count. SCEV returns 0

Hi Everyone, I am testing Asterisk 1.8 AMI events. The voipinfo page on AMI events is specific to 1.6. I am wondering if the developers cared to write about the new events that are spit out in Asterisk 1.8 somewhere on the web? I checked the tar ball for asterisk 1.8 and documentation doesn't include this event: *Event: Unlink* Privilege: call,all Channel1: SIP/9999-00000029 Channel2:

I want to build my own calling card system on Asterisk. I looked at this page - http://www.voipinfo.org/wiki/view/CallingCard+Applications and it has listed some applications that I thought could help speed up the development process though the link down the bottom doesn't work. Does anyone know of any AGI etc applications to build a Calling Card system on Asterisk? Michael

I have an * installation which will refuse incoming callers once a max (5 callers) is reached. Caller 6 and up should be notified of congestion...without network load on my trunk. How would I do this? The voipinfo wiki shows playing a congestion tone to the caller, but that seems stupid since I'm consuming bandwidth to send a tone. I also tried just responding with the congestion

Markus Weiler <markus_weiler at mailworks.org> schrieb: Hi > from voipinfo... > > If an Asterisk command specifies a sound file in a*subdirectory*, > Asterisk looks in that subdirectory for the language subdirectory. For > example, theSayDigits > <http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigits>command may > play the sound file

Wow, that sounds great! Thanks for working on this, and yes, please, send the patches! --renato On 30 April 2013 16:23, Murali, Sriram <sriram.murali at intel.com> wrote: > Hi Daniel,**** > > I made some changes to the LNT perf reporting tool to make it more user > friendly by adding some features:**** > > **1. **Make the sidebar and the navigation bar stationary,

Hi Sriram, Thanks for performing this analysis. The problem here, both for memcpy and the vectorizer, is that we can’t predict the size of “n”, even though the only use of ’n’ is for the loop bound for the alloca [4 x [8 x i32]]. If you change the unroll condition to TC >= 0 then you will disable loop unrolling for all loops because getSmallConstantTripCount returns an unsigned number. You

Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the

I just want to pdf and email faxes coming in over pstn on a TDM400P. Outgoing faxes would just go out over pstn, not through asterisk. All the voipinfo , etc, howto's are quite complicated. And most use third party apps like Hylafax. I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm now using 1.4.22, but I'd go to 1.6 if it made this easier. But I've