similar to: Playing music while transfering

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

Actually it worked, but only after I defaulted all the settings on the phone and let it pick the config up fresh. Anyone know if there is any headset config options to default to headset/speaker? Thanks Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Lee Archer Sent: 03 January 2006 14:49 To:

> -----Original Message----- > From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] > On Behalf Of Chris Lattner > Sent: Friday, March 18, 2011 1:15 PM > To: Anton.Lokhmotov at arm.com > Cc: llvmdev at cs.uiuc.edu > Subject: Re: [LLVMdev] [PATCH] OpenCL half support > > > On Mar 17, 2011, at 10:17 AM, Anton Lokhmotov wrote: > > > Hi

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

I have set up a * box with two hfc ISDN pci cards using mISDN both in TE mode with PmP mode. (using $MODPROBE hfcpci protocol=0x2,0x2 layermask=0xf,0xf) I have no problem dialing out by explicitly naming the mISDN port, ex: Dial(mISND/1/${EXTEN},60) or Dial(mISDN/2/${EXTEN},60) But it does NOT work when specifying the mISDN group: exten => _(outpattern),1,Dial(mISDN/g:TEmode/${EXTEN},60)

I was having video problems, so upgraded my Linux box from SUSE 9.3, where LLVM frontend 4 source built fine, to SUSE 10.1, where I got the error message: ../../llvm-gcc4-1.8-source/gcc/libgcc2.c:541: internal compiler error: Segmentation fault Please submit a full bug report, with preprocessed source if appropriate. See <URL:http://llvm.org/bugs> for instructions. This version of SUSE

Hello Happy New Year! I succesfully installed the Ports of Zaptel BSD 1.4.0 and Asterisk 1.4.13 (that's the latest in the Ports). To save CPU, I'd like to play PCM WAV files instead of eg. GSM. Per... www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk ... I recorded a sample of my voice using XP's Sound Recorder, then ran the following : sox test_wav.wav -r

Hi, I'm testing/debuging the migration from mbox to maildir in a mailserver running Dovecot 1.0alpha5. I migrated all 472715 emails(in a test server) and later noted that dovecot could not get the same flags/status of the emails as in the mbox. I was using perfect_maildir.pl and I think its not dovecot fault. Seems to me that perfect_maildir.pl convert the emails in a way that dovecot could

Dear Mr,Ms; I am planing for a custom IVR, for example to act as a simple installer! I mean there is some choice via 0-9 and # as *Next* and * as *Back* button. is there any way for me to detect if the caller pressed # vs * on Dialplan ? -- Regards, Ali R. Taleghani 0936 322 4069 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Robert, On Mon, 2006-11-06 at 12:45 -0800, Robert Mykland wrote: > Reid, > > Here's the backtrace you asked for: > > (gdb) bt > #0 0x0862d65c in llvm::LiveVariables::runOnMachineFunction () Hmm, this is a little strange. Your LLVM build is non-debug (there's no line numbers or arguments in any of the llvm related calls). However, your llvm-gcc build seems to have

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

On Mar 17, 2011, at 10:17 AM, Anton Lokhmotov wrote: > Hi Chris, > > So what do you think about this proposal? If you agree, it would be good to > include the patch into the 2.9 release (to avoid breaking compatibility > later). Hi Anton, I'm sorry I don't have the patch anymore. Please resend. It is too late for new features in 2.9 though. >> The last paragraph

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

The HAVE_MMAP_FILE macro comes from when you configured llvm. The configure tested if your system is giving access to a various resources required by llvm. LLVM needs the MMAP functionality by your system. Some times, if the required resources aren't present, we use other resources. However, in this case the building process is stopped, because the functionality can't be implemented

On Mon, Apr 7, 2008 at 5:56 AM, Vania Joloboff <vania.joloboff at inria.fr> wrote: > We do dynamic binary translation. We are in a similar situation to qemu > except we are SystemC / TLM compliant for hardware and bus models. Our > current technology is somewhat like qemu, we translate the binary into > "semantic ops", which are pre-compiled at build time, like qemu.

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: