similar to: can we lend a hand?

(forgive the brief interruption to -users with a mostly -dev issue, just wanted to publicize this on behalf of the larger community) If there are any ambitious coders out there (not too many shekels yet, but I expect some folks may pony-up) please see: www.voip-info.org/?page=Asterisk+Bounty+Email+TTS We are at $150 & counting. Maybe lobby your exec's for $50 to contribute to this,

I have PolyCom phones in one office working perfectly, but in another office with a new subnet, new server, new everything, the time does not work. Everything else about the phones seems fine, but the time. If you look at the internal webpage in the phone, it shows "clock". Our server, which is configured to allow others on the net to get their time from it, and it in turn gets its

All, Just came across the prompt #3 from inside the top menu of VM in latest stable. Allison does not announce the prompt, but if you know it is there, you can press 3 & successfully follow the prompts from there to send your message to other users on the system. But, of course, obviously, I am asking: how do I resolve the situation whereby the users are not hearing this prompt? (since most

Full_Name: Christophe Genolini Version: 2.8.1, but also 2.9 OS: Windows XP Submission from: (NULL) (82.225.59.146) I am using matplot with the option lend="butt", but only the first line (the black) is printed correctly : > matplot(matrix(1:9,3),type="c",lwd=10,lty=1,lend="butt") Gabor Grothendieck find the problem in matplot code: the ... is passed to plot

Comrades, This is somewhat off-topic, but I hope also somewhat on-topic, since I'll bet a lot of others could save some time if this was able to be pulled-off. (in fact, I think it is, but I don't know where to begin) Is there a way to initialize my freshly downloaded & installed updates to the kernel & kernel source, without rebooting? I pretty much use either SuSe or CentOS

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Hey, Hi, All, We have a few dozen of the Kirk (ie: Polycom bought this European brand) out there & most all work very well & work very well with most all versions of Asterisk. But we have been tripped-up by one combination of firmware & version & configuration variables. We are running Asterisk 1.4.23.1 (TrixBox CE). We are running latest stable firmware on the handsets. Most

How can it be, that class 1:3 in my case borrows, when all sibling classes are isolated ? nessus:~# tc -s -d class show dev eth1 class cbq 1: root rate 100Mbit cell 8b (bounded,isolated) prio no-transmit/8 weight 100Mbit allot 1514b level 2 ewma 5 avpkt 1000b maxidle 1us Sent 484 bytes 7 pkts (dropped 0, overlimits 0) borrowed 0 overactions 0 avgidle 77 undertime 0 class cbq 1:1 parent 1:

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor

I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. Thanks. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

HEllo list ! When user A calls user B via Asterisk (Users A and B are registered on the same Asterisk server ) and an ISDN PRI, user B phone always shows Asterisk server telephone number. How to hide it and how to forward user A number ? We tried usecallerid, callerid, hidecallerid, restrictcid, usecallingpres in zapata.conf but we always see Asterisk server telephone number ! Thanks

I think I've forgotten something obvious.... I've got 2 incoming calls, I want to bridge them - how can I do this ? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital.

Joyful. I'm adapting a FORTRAN 77 package for use with R. Pretty straightforward. Except for a glitch it took me some time to figure out. This existing package has subroutines which have parameters called "NA". So, I called subroutines like bnodes <- function(n, lst, lptr, lend, nodes, nb, na, nt) { ensure.all.numeric(list(n, lst, lptr, lend, nodes, nb, n.a, nt),

Hello everyone. For reasons too long to explain I wanted to do plots similar to histograms with plot(type="h"). I ran into a problem - if I set line width too high, histogram isn't accurate anymore. For example: par(lend=2) plot(c(2,4,3,2),ylim=c(0,5), type="h") abline(h=3) Column 3 appears just as high as it should. But if I do par(lend=2)

Hi, I am trying to plot odds ratios and the corresponding confidence intervals in horizontal segments. It would be ideal if the confidence interval segment can be drawn with little vertical bars at both ends. I have tried very hard to change the type of ends by using 'lend' arguments, but cannot make it. I even tried 'arrows()', but still failed. Following is the code I use:

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 -