similar to: Query regarding connecting PABX with Application server

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi Folks, I am running a SPA-3000 behind a legacy PABX on an analog line. I have been able to set up a dial plan that sends outgoing calls out to the appropriate VSP depending on prefix, and that part and the incoming call handling works fine. I am now trying to implement call pickup (dial 6*) or manual call forwarding (flash, dial extension). On the first of these I have worked out how to get

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall.. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 1111112222333333444444 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil _______________________________________________________

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

Hi all I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk 1.8.11.0. The reason is the customer has a large installed base of Samsung phones physically connected to it and on each users desk. They did not want to spring for a complete replace of all their Samsung phones with generic, and

Hi all, I've been lurking here and reading the Wiki for a month or so now, getting information on the suitability of Asterisk for my installation. I'm responsible for the technical stuff at a mostly-inbound call centre in Melbourne, Australia. Due to our rapid expansion, it's getting close to the time to put our old beloved Fujitsu 9600 PABX out to pasture. I'm evaluating

Hi guyz I m very much new to Asterisk and i want to establish a PABX for my house at the moment i have X100 card(Configured) Asterisk PABX(Installed) Now i want to know how would i recieve call for e.g. i want during incoming call (after demo mess). if user press 9 (for e.g) the call redirect to host1(for e.g. again).or if ne user of gnophone want to dial outside how would he do so . N e sort

Hi all,as good? It would like to know if already they had had success, in the integration of the functions of asterisk, with one pabx legacy (Avaya)for that pabx avaya, could use the voicemail of asterisk. For sample, user of pabx avaya, it would have its calls directed for not attendance and busy, for asterisk and asterisk, it would send the same one for the voicemail. Best Regards Josu?

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Good morning all, I have a question: if I want to connect an * to a traditionnal PABX via TDM04B to get PSTN line (incoming and outcoming call), are FXO enough (no analog phones connected to *)? I just want to add PSTN call (in and out) possibility for SIP or IAX UA. From what I saw here http://www.voip-info.org/wiki-FXO I'm right. Thanks. -- Daniel

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks.

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

Dear list, I am still struggling with something that should be easy: I compare two data frames with a lot of common rows and want to keep only rows that are NOT in both data frames, unique. Here are an example of these data frame. reported <- structure(list(Product = structure(c(1L, 1L, 1L, 1L, 2L, 2L, 3L, 4L, 5L, 5L), .Label = c("Cocoa", "Coffee C", "GC",

Hi, say I got the function: > x=function(nbr){y<-rnorm(nbr);y1 <- mean(y);plot(y)} how can I retrieve value of y1, when I need it. I don't want: > x=function(nbr){y<-rnorm(nbr);y1 <<- mean(y);plot(y)} > y1 I want someting like: "x$y1" and then I get the value Many thanks, robert -- View this message in context:

Dear List, I want to compare and return the rows which are NOT in the two data frames. Classic methods don't work as the df have NOT the same dimensions. Here are one example of my df: reported <- structure(list(Product = structure(c(1L, 1L, 1L, 1L, 2L, 2L, 3L, 4L, 5L, 5L), .Label = c("Cocoa", "Coffee C", "GC", "Sugar No 11", "ZS"),

Stephan, Ok, I'll re-state the problem... I have two devices that I want to talk to each other: 1. an Asterisk PBX 2. a Damm Cellular TETRAFLEX digital radio system (www.damm.dk) both devices are effectively "gateways" because they have many subscribers behind them. The Damm Cellular system controller is based on Windows-XP Embedded and its sub-systems used the OpenH323

In lattice, using the command trellis.par.get for superpose.symbol, plot, symbol and/or dot.symbol shows that we can specify alpha, cex, col, fill (for superpose.symbol and plot.symbol), font, and pch. Trial and error shows that the font affects letters but not pch=1 or pch=3 (open circles and plus signs.) I want to use open circles and plus signs, keep the colors and cex I've specified but

Overlap receiving timeout, plus dialplan latency, causes network to retry SETUP https://issues.asterisk.org/view.php?id=16789 This patch removes the requirement that some may have found that you need to insert a Proceeding() statement very early in your dialplan, otherwise an inbound overlap call may retry and fail. Our experience was from a PRI connected PABX, if we took too long doing