similar to: Corporate Feedback to OSS (was: Re: Mushtaq Ahmed is out of the office.)

I will be out of the office starting 07/06/2007 and will not return until 07/09/2007. For anything urgent pls. contact Fritz Reeve. CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is being sent by 3Com for the sole use of the intended recipient(s) and may contain confidential, proprietary and/or privileged information. Any unauthorized review, use, disclosure and/or

Hi, Does anyone have a binary of XML package for 1.7.0 on Windows? I have searched it at the RSXML pages (http://www.omegahat.org/RSXML/ ) but it seems it is no longer support. Any pointers for the compiled version will be appreciated. Regards, mushtaq [[alternate HTML version deleted]]

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

Hi: Does Trixbox support ACPI and IO-APIC out of the box? My Trixbox server isn't seeing the mainboard's APIC. -Stephen-

It is my understanding that R is licensed under the GPL with the exception of a few header files for the purposes of linking binary code with R under non-GPL licenses. However, the R-base package itself is licensed under the GPL, as are many (but not all) packages in CRAN. Furthermore, basically any R script will use functionality from R-base. As I understand it, the situation isn't

Hi, everybody: As part of a paging macro I'm using SendDTMF to send digits to the called party. The section looks like this: exten => s,1,Wait(0.5) exten => s,n,SendDTMF(9531290) exten => s,n,Wait(1.0) exten => s,n,Set(MACRO_RESULT=CONTINUE) To test I direct the call to a live extension just to hear what's happening -- what actually happens is that only the 9 is sent, and

Hi, folks: I need to be able to have a single DID ring multiple remote (IP and PSTN) extensions, and then pass the call to whichever picks up first. I'm sure this is old hat -- lots of providers offer it. I see that Trixbox will do it, but it's not clear how it's doing it. They use different terminology -- a "ring group" and "hunt strategy" How can it be done

the caller is out his money anyway when you call any phone and voicemail kicks in, although i think on a payphone they give you a 2 or 3 second window to hang up. Suggest you implement i'm here / i'm away dialplan logic or set the do not disturb button that way when someone calls and the guy is away it hits voicemail right away and the caller can hear this and still have the 2 or 3

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Hi: Say I want to build an IVR application which sends an SMS message to a mobile telephone when the caller responds to a prompt in certain way. I think I can manage the part about generating the message and building something to actually send it. The part I'm foggy about is: how would I actually get the SMS message to the carrier? Discussions with the carrier have led absolutely nowhere

i586 wine versions for opensuse once had a 32 bit download that accompanied the i586 wine version. It no longer does. I haven`t been able to get wine running my apps since then. Do i need to start using the 32 bit listed in x86_64? I`m lost please help. I`m stuck with older wine versions until I fiqure this out.

Hi All, I would appreciate a lot if you could help me. I have installed Asterisk 1.4.1 and zaptel 1.4.2 on my redhat enterprise linux 4. I have also installed 1 FXO port card: Digium TDM400P. After loading zaptel driver I could see my digium card's led glow green. Tested with zttool that its in OK state. I have configured fxsks=4 in zaptel.conf(channel 4 cause FXO module is on port 4). I

test only. good luck! james.zhu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070807/0fd2b827/attachment.htm

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

Hi: This should be easy. I'm running 1.2.15. When a caller calls someone's voice mail, it goes straight to a beep, even though there is an unavail.wav file in that user's voice mail directory. Here is the relevant part of extensions.conf: [internal] exten => 2211,1,Dial(SIP/211,10) exten => 2211,2,VoiceMail(u211@default) exten => 2211,3,Hangup Here is the relevant part of

Hi, Anybody lucky with this config inside an Asterisk server for dealing with FAX ? FXO_LINE ----> ASTERISK 1.4.2 ---> IAXMODEM ------> HYLAFAX TDM400P ZAPTEL 4.3.1 1 FXO port 1.41 ------------------------------------------------------------------------ I know Fax is not officially supported on TDM400P cards but I did not expect not being able of sending one

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 On Wed, 2023-02-22 at 11:08 +0000, Marc wrote: > I don't even get what the advatages are of doing this with sql. If you > use local replicated ldap and use local credential caching then your > master ldap can go down without issues, even the local caching handle > some local slapd issues. Going to have to +1 this. LDAP also does

Hi, everyone: I'm in the processing of deciding what IP phones we should use with our Asterisk server, and I wanted to get feedback from the user community on the quality, reliability and ease of operation of Snom phones. What do you have to say about these phones? Are there other phones you'd suggest along with or instead of Snom? Thanks, -Stephen-