similar to: caller ID strangeness

Hi there! I remember reading that you cannot reload new zone files on the fly and require a full restart of the nsd daemon? We are evaluating multiple DNS servers that have better performance comparing to bind, but will require quite heavy zone reload (new and existing) every 10 minutes or so. Downtime (even 1-3 secs) is not the option. Thanks!

Hi I have tried to setup freebsd (with samba3 ports), backend is openldap with smbldap-tools. After a windows client joined the domain, I tried to right click a folder to set Security/permission. The windows client has a popup windows which display: The program cannot open the required dialog box because it cannot determine whether the computer named "DOMAIN" is joined to a domain I

*@home has this for it's incoming fax macro --- start snip --- [ext-fax] exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1) exten => in_fax,2,Macro(faxreceive) exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf ${FAXFILE}.pdf) exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from ${CALLERIDNUM}

Hi, I would like to know if the following is possible: * how to accept only one call at a time on a particular SIP extension (softphone). I'm referring to incoming calls. Can it be done on the server side or just on the client? ie. all other incoming calls will just be dropped while the extension is busy. In other words I would like to simulate having just one phone line available. I tried

I've some code to manipulate incoming Caller-ID - so its suitable for replying to... [sipdef] exten => s,1,NoOp(FWD SIP: "${CALLERIDNAME}" <${CALLERIDNUM}>) ; Alter incoming calles from pulver - add a '87' exten => s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4) exten => s,3,SetCIDName(87${CALLERIDNUM}) exten => s,4,SetCIDNum(87${CALLERIDNUM}) exten

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten =>

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

it actually is complicated. the libFLAC api is not suited to a multithreaded design because the i/o is stream-based, not file- based. flac(.exe) is the file-based wrapper around libFLAC that allows it to work on files. the way libFLAC buffers data is also impossible to parallelize without significantly changing the api. it would take a specialty file-based encoder using an independent frame

On 9/6/07, Harry Sack <tranzedude@gmail.com> wrote: > it's really not complicated I think: only api changes to write on any Please get started writing a patch, immediately. -- avuton -- Anyone who quotes me in their sig is an idiot. -- Rusty Russell.

Hello All, I was struggling with the need of creating flac files over 2GB recently. Both the command line flac and my util using stream encoder returned FLAC__STREAM_ENCODER_CLIENT_ERROR when reaching that point. Finally I made a small change in libFLAC/stream_encoder.c in file_tell_callback_ function. I replaced the ftello call to fgetpos. This seems to work very well under win32. Based on

I just read on Gizmodo that the new Denon AVR-5308CI will include FLAC decoding! According to the text, it supports "USB inputs for bus-powered HDD, along with AAC, FLAC, WMA Lossless, MP3 and WAV decoding" Very happy to see FLAC being supported by a big name like Denon! Josh (or anyone else), curious if you know to what extent the support is offered or if you have any further

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/9/8, Josh Coalson <xflac@yahoo.com>: > > > > it actually is complicated. the libFLAC api is not suited to a > > multithreaded design because the i/o is stream-based, not file- > > based. flac(.exe) is the file-based wrapper around libFLAC that > > allows it to work on files. the way libFLAC buffers

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels

2007/7/3, Ralph Giles <giles@xiph.org>: > > On Tue, Jul 03, 2007 at 07:18:17PM +0200, Harry Sack wrote: > > > Can anybody tell me what the maximum filesize is for an image file to be > > included in a FLAC file? I'm just exploring this feature and I didn't > find > > this yet > > The metadata block length field is 24 bits, so the upper limit is

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

Hey, all. I'm trying to set my CID such that, internally, I see a four-digit extension (which is also handy when checking VM), but externally, I see the full 10-digit number. So I plugged these lines into my extensions.conf: exten => _XXXXXXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2) exten => _XXXXXXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1}) exten =>

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various

Is what I am trying to do in this context possible. That is changing the incoming CALLERIDNUM. In this case if the incoming CALLERIDNUM is not preceeded by a "1" I want to add a "1". Often calls come in without the preceeding "1" and this plays havoc with my redial if the 3 digit area code matches a local 3 digit extension. All my outside calls are 10 digits or 1+10