similar to: Call interruption

Hi all On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? Thanks! Andre -------------- next part -------------- An HTML attachment

After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yuan Liu

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Dear all, Some functions like 'ROC(Epi)' can be used to perform ROC analyssi, but it needs us to specify the fitting model in the argument. Now i have got the predicted p-values (0,1) for the 0/1 response variable using some other approach, see the following example dataset: id mark predict.pvalue 1 1 0.927 2 0 0.928 3 1 0.928 ..................

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

Hi all On the Asterisk website in the blog its announced that in a next release Asterisk would support dynamic DUNDi weitht values. I've installed Asterisk 1.4.4 (via aptitude install) but this doesn't seem to work. Has somebody some experience with this or know whether this feature is already implemented or not? Thanks! Andre -------------- next part -------------- An HTML attachment

Dear List, I'm having considerable trouble setting up my environment (Linux, Fedora 16, Bash) to build and check packages under R Under Development (r61660). I'm doing this to better get a handle on difference in the output from running checks on examples in one of my packages. Note I compiled R Under Development myself The problem I am now having is whenever I try to build my analogue

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

Hi All I've done some reading on the wiki and read some of the mailing list archives, but can't see anything on this. I guess this means I'm either searching on the wrong thing, or have totally the wrong idea... Can anyone suggest if the following is possible? Currently, our office has a 24 analogue extension PBX, and 2 ISDN lines providing it with external connectivity. We have

hy all actually i want to have a setup of five offices having round about 200 extensions ( each office having 35 to 45 ) which will be connected through asterisk. now either i should go for voip phones( hard phones ). or use any interface card to asterisk server to which the analogue phones will be connected. ----- if i use analogue phones in the above case ( we have analogue phones already )

I am a newbee to asterisk. I recently installed asterisk@home. Everything went well and my set up is running fine with soft phones, such as kphone and XtenLite. Now, i want to be able to connect my analogue phones to my asterisk pbx box and use it as if i make a regular Phone call (I do have my PSTN gateway account with broadvoice.com and already configured to route through it). I do NOT have a

Hello; I am facing a trouble with A2Billing when using analogue lines because the channels are not closing properly when dialing happen through A2Billing (it seems the dialing scenario including the hangup is not handled properly through A2Billing but I do not have control on this). But when I do dialing from asterisk and using analogue lines, I do not face a trouble because I can write the

So made my first forray into 1.4 and DAHDI and hit a problem. (Not convinced this is a DAHDI issue though...) Testing an analogue line and asterisk sees the caller ID being passed, but then fails to detect ringing. A plain old analogue phone plugged in rings just fine. Console output: == Starting post polarity CID detection on channel 4 -- Starting simple switch on

Hi all, the company I work for has an aging Digital PBX attached to an E1. This PBX has a few analogue lines, one of which we use a 'traditional' fax machine on. I want to upgrade our PBX and Asterisk is almost a perfect fit. The only problem I can't seem to find a working solution for is Faxing. I don't want to use Hylafax or other similar methodologies. I believe there

I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it

I try to perform a clustering using an existing dissimilarity matrix that I calculated using distance (analogue) I tried two different things. One of them worked and one not and I don`t understand why. Here the code: not working example library(cluster) library(analogue) iris2<-as.data.frame(iris) str(iris2) 'data.frame': 150 obs. of 5 variables: $ Sepal.Length: num 5.1 4.9 4.7

I have a potential customer who has an existing PBX with analogue FXS ports connected to phones. He wants to allow a single remote worker to be connected to one of the analogue extension ports using VOIP. I know I could do it using Asterisk with an X100P card, but that seems a bit overkill. Does anyone know of an analogue->VOIP adapter that has an FXO port in it instead of just an FXS port?

Hi, I am running Asterisk v1.2.7.1 with a Digium TE110P. My dialplan is very simple, when a call comes in on my analogue BT PSTN line, it rings the other ZAP interface (my house phone). Slightly pointless (having a 1x1 switch) I know, but I am planning on doing more with internal SIP extensions, and outgoing SIP services etc.... At the moment it's just simple though whilst I try to fix this

In Australia, Telstra, the local telco provides isdn modem for isdn connection. The modem has 2 analogue telephone jacks and a serial port for connection to dialup internet. My question is that will it be possible to use Zaptel TDM02B to connect to the analogue jack instead of getting a fritz card to do the telephony. Will there be less feature if doing so? -- David Kwok, CISSP Tel: 612

Hi, Firstly, I'm new to Asterisk and am a system admin rather than a phone engineer. I've googled and read around but haven't been able to answer my questions sufficiently to buy hardware and get this thing set up. Secondly, if I've missed vital information from what is below, please let me know what. Onward...: So, what I'm trying to solve us remote working. We're a