similar to: Caller ID info ZAP --> SIP??

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

I have a habtm relationship like the following: Automobile <--> Options What I want to do is when an option is deleted, make sure that there are no Automobiles that still have that option. It seems simple enough, but I am having a tough time with it. I believe Rails automagically deletes the relationships when the automobile is deleted, but not the other way around. Any help would

Hi All Total noob on the list so all help appreciated.... I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows). I've plugged in two Cisco 7960 phones.... The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......

I just installed Wine, and when i test it by the command "wine cdplayer" i get the following error message : "err:module:BUILTIN32_dlopen failed to load .so lib for builtin kernel32.dll: symbol __sysconf, version GLIBC_2.2 not defined in file libc.so.6 with link time reference err:module:fixup_imports Module (file) KERNEL32.dll needed by C:\Windows\cdplayer.exe not found" I

I am running (testing) a noob-tube for myself as follows: Windows XP SP2 lighty ruby 1.8.x rails 1.1 scgi I have everything working fairly well, but with one minor exception: the actions that I have defined as follows are not found. class MytestController < ApplicationController def index render_text "Hello Nurse!" end def dilbert render_text "Dilbert

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47

All this talk about vorbis comments has made me think. Would it be possible to create backups of the Vorbis comments for my vorbis collection (currently about 1850 files) ? Instead of having to make sure I enter the correct details into CDex (saving in to cdplayer.ini & backing up that), or if I have comments in the Vorbis files that aren't stored in cdplayer.ini, such as the tags

Hi there, My name is Angelica and I was just checking out your page at http://us1.samba.org/samba/docs/SambaIntro.html and noticed that the the link to the Dilbert site on http://www.unitedmedia.com/comics/dilbert/ isn't working. I'm a Dilbert fan myself :) so I thought I'd mention a page I have bookmarked: http://www.answerconnect.com/articles/dilbert-fans It has a bunch of

Dear Opus family, we have implemented the Opus codec at a Texas Instruments DSP C6418. It is working fine! Does anyone has experience with the configuration of the codec for a speed optimized implementation on that DSP? At the moment, we use the following settings: #define NONTHREADSAFE_PSEUDOSTACK 1 #define FIXED_POINT

Hi Jean-Marc, thank you for that suggestion! It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option. Maybe it was only part of an early version of the Opus sources? I looked for the file in versions V1.1, V1.1.1, V1.2alpha and V1.3 but did not found it. Do you have an idea, where I can get the

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

Since I have a mixture of RH 7.1, 7.2, 7.3 and Advanced Server 2.1 boxes, I've made a set of RPMs for those platforms. I do not have a RedHat 8.0 box to make RPMs. Since RedHat <= 7.2 is out of support by RedHat, I don't think we'll see any officially created RPMs (but I could be wrong about that) so I thought I'd offer these to the general community. These were all built

On Mon, Jun 04, 2007 at 03:20:05PM -0600, Peter Saint-Andre wrote: > >>Hi Luca, > >> > >>Thank you for working on the Theora and Vorbis drafts in the IETF's AVT > >>WG. Do you also plan to work on the Speex draft? > > > >other people are working on the speex I-D, Ralph should be able to tell > >you which to contact ^^; > > Hi Ralph,

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

Hi all, Apologies in advance, this email is quite long. I've prepared an updated Vorbis RTP Internet Draft, which is a continuation of draft-moffitt-vorbis-rtp-00.txt which can be found below. If this new draft gets the ok I'd like to submit this to the AVT WG later this week. There are a number of changes over the original I-D, notably the changing of the M bit function in the RTP

Sorry, just one more thing I've just noticed now from looking at the IMAP session IDs: Dovecot seems to map the steps as outlined below to four different IMAP session IDs all related to the exact same Thunderbird client instance: * <yH0lKWRRpNHAqOh5> session for the folder copy of folder "Dilbert" with 317 messages * <VcMlLWRRqNHAqOh5> session for the deletion

That log didn't do much help, can you try enabling mail_debug=yes for a moment and see if it gives any more insight? Aki On 05.06.2017 15:13, awl1 wrote: > Hello again, > > here you are (log excerpt from May 29 00:06:02 until May 29 01:17:49): > > May 29 00:12:05 imap(xxx at xxx.org): Info: IMAP session state is > inconsistent, please relogin. in=1369 out=222662 > May

Hello all, I am writing a "virtual cdplayer" application, that plays tracks from my cd backups, which are stored as a .flac file and a cdrwin style .cue file. Anyway enough of that, onto the problem.. I wish to use C++ for my program, but when I try to compile it with g++ I get the following types of errors: /home/dave/development/splitimg/splitimg.cpp:153: undefined reference to