similar to: New app: Consultative transfer for each phone

Hi Stefan, > At the moment I'm into the Bleuz mailing list for getting the SCO > feature to work, starting with my laptops BT and my dads Nokia. > > I have now a basic understanding from the RFCOMM layer which delivers > the AT commands, AT works. If I get the basic SCO functions > working with > the test tools, to get the headsetfunction itself into the > pc.

In case others find this of value, I wrote a patch to rsync 2.5.6 to give rsync in --daemon mode the ability to run a pre-transfer and post-transfer command. These options handle our need to prepare a server to receive files and to do some processing after receiving files. The options for /etc/rsyncd.conf are pretransfer script = /some/command/to/run posttransfer script =

Ist there any possibility to use the funktion "consultative transfer"? ( have 2 ISDN-pones attached to the hfc-nt card, configured as zap) With the "#"-key it ist possible to park the call or to make a "blind transfer" at the moment. I have activated threewaycalling in the zapata.conf file: ; internal S0 bus (first hfc/s card): context=local signalling =

I'd like to monitor key press in a meetme room. Is it possible when connecting one side of a local channel in the meetme room and the other side of the local channel to an agi with the command "wait for digit" ? Thanks Eric

hello all- Long time listener, first time caller. This is a great list and has given me tons of help as I've set up * for the first time. I've got an asterisk system up and running at a new company, and it does about 99% of what we need it to do. TelephonyWare has been our equipment supplier, and has been great with support, but I've got an issue that has us both stumped.

Hi all, The following code snippet isn''t working terribly well for me. I can get the Bias, StandardName and DaylightName, but everything else is goofed up. The alignment seems ok, but maybe I''ve missed something or maybe I have to do extra work to unpack the SYSTEMTIME structures. require ''windows/time'' include Windows::Time buf = 0.chr * 172 #

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Ok. I've noticed a thing: when you ring a sip phone, and hangup before it answer, asterisk sends a CANCEL to the phone to abort the current operation (in this case, the INVITE). and this's correct according to rfc. But now... when a sip phone A is ringed from a phone B , and that call from B is picked up by the phone C via *8 , asterisk sends 'BYE' to the phone A ( C & B are

Hello all, I have a population of 112 servers that are experiencing different levels of packet loss. I don't want to poll all 112 of them (the analytical tools must be manually run on each individually) so it seems best to sample among them; then I plan on using R to run comparisons of the data pulled from each one. I'm not clear on the most sound way to go about this and I don't

Hi, I am trying to find remote call forwarding feature in asterisk. I don't know is it possible or any one had already done it. SBC (local Telco) provide such feature. I can call into my voicemail number, and set the remote-call-forward to my cell or another number. It is like person can remotely manage to set the call-forward or DND to his/her extension. Can this be doable in asterisk?

Hi all. I have and E1 channel bank from Loop Telecom. there's a little issue with it, I cannot ring the phones on fxs interface, but can connect without issue them. What happens: I dial the phone on port 1, asterisk says "Zap/1 is ringing", but the phone on the analog port doesn't ring. but if I take off hook the ringed phone, asterisk detects the answer at they're bridged

I have a strange problem when calling some numbers with asterisk, I get an hangup for busy condition even if the phone at the other end isn't busy. I can route the calls via SIP to another carrier and then I have a SIP code 486 or I can terminate them on digium cards (E1) and I have an Hangup code 17 I know for sure that one of the numbers is hosted by a different provider than the one

I write about R every weekday at the Revolutions blog: http://blog.revolution-computing.com , and every month I post a summary of articles from the previous month of particular interest to readers of r-help. You can find older summaries at http://blog.revolution-computing.com/roundups . (By the way, the blog celebrated its first anniversary in December. Blame the celebrations and the holidays for

Dear all, I would appreciate please a piece of help regarding the use of acast/dcast functions in reshape2 package. Specifically, I'm working with a data frame, that has information about SAMPLE, GENE, and TYPE of MUTATION (as shown below): Sample Gene Type 22M AEBP1 SNV 17M AEBP1 SNV 22M ATR INDEL 22M ATR SNV 11M BTK SNV 11M BTK

Dear all, as the bug # 7991 is flagged not-reproducible, let me give you some pieces of code, as I have the same or similar problem. The problem always shows up with the first example (a small text file) and only sometimes (but without obvious pattern) with the second example, which is a binary file. > download.file("ftp://ftp.nhc.noaa.gov/pub/atcf/btk/bal012006.dat",

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Hi Do you want something like dcast(test, Sample~Gene, fun=function(x) paste(x, collapse=",")) or dcast(test, Sample~Gene, fun=function(x) sum(as.numeric(x))) 1 means INDEL, 2 means SNV and three means both Cheers Petr > -----Original Message----- > From: R-help [mailto:r-help-bounces at r-project.org] On Behalf Of Bogdan > Tanasa > Sent: Monday, July 3, 2017 9:22 AM

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

Good day all I got my Music on hold to work but can I/how do i get music on transfer? Please help Thanks

Good day all I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added "exten => 403,hint,SIP/403" in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? Please Help Altus