Re: [asterisk-users] AsteriskNOW Web GUI

Steve Totaro wrote: > Awesome, when you say end user do you mean the people sitting at the > phones or the person doing MACs, or both? people at the phones: yes the person doing MACs: what is that? (your obviously not talking about the one sitting on my desk :) Philipp -- amooma GmbH - Bachs

Re: [asterisk-users] AsteriskNOW Web GUI

http://en.wikipedia.org/wiki/Gemeinschaft ) It *does* have a web based end user GUI. But "end user" does not include the admin. So the MySQL replication or cluster needs to be set up by hand - although we provide them with a detailed step by step manual. Regards, Philipp Kempgen --

Re: [asterisk-users] Distributed System

tware in about 3 months or so (can't promise that, don't nail me down to it) under the GNU GPL. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http:/

Re: [asterisk-users] Is it possible to register without sending the password

.conf Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Geschäftsfüh

Re: [asterisk-users] Distributed System

Seysan wrote: > Is there anywhere that we can look into for Realtime + MySQL that you > mentioned? Maybe http://www.voip-info.org/wiki/view/Asterisk+RealTime http://www.asteriskguru.com/tutorials/realtime_pgsql.html Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 N

Re: [asterisk-users] Distributed System

ine if you are doing "home user" > type stuff where you don't have lots of users within one context, but if > you have offices with several people, I just see lots of potential > issues. I could be wrong, but I've never been able to figure out a way > around it.

[asterisk-users] Zaptel hardware for timing (was: Re: app-conference)

ram wrote: > app_meetme can use ztummy but on highload expect to use hardware source A thing that was on my mind for quite some time now: Would it be beneficial to have a Zaptel compatible card in a system just as a timing source, even if it's not connected to a PRI? Regards, Philipp

Re: [asterisk-users] PRI cards, Digium vs. Sangoma

d kernel > panics when trying to shut down wanrouter. I don't have this 'fear' > with Digium cards. So one could easily look at it the other way round and say that once you've made sure there are no interrupt issues and got the Digium card up and running it won't b

Re: [asterisk-users] Is it possible to register without sending the password?

al_sip] > and the endpoint username was "bilal_1000" then I see > a the message (log) that declare that registeration > from bilal_1000 failed (ofcourse because bilal_1000 is > not configured while bilal_sip is configured in the > sip.conf). Could you send the part of yo

Re: [asterisk-users] Restart status

tly monitor asterisk -rx 'core show uptime seconds' And if that does not drop to let's say less than 30 seconds: asterisk -rx 'stop now' sleep 1 killall -9 asterisk asterisk Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.am

Re: [asterisk-users] which OS would be fine for asterisk

ith YaST they'd really mess things up - without even knowing. I'm so glad nobody in this thread has argued for using Windows. ;) (It doesn't even come with an ssh client! You really feel like your hands are tied.) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Ne

[asterisk-users] PRI cards, Digium vs. Sangoma (was: Re: Saftware RAID1 or Hardware RAID1 with Asterisk)

a Digium card? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Gesc

Re: [asterisk-users] Asterisk 1.4.11 released

The Asterisk Development Team wrote: > The Asterisk development team has released version 1.4.11. > This release is available for download from > http://downloads.digium.com/pub/telephony/asterisk/. Not quite. :) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 5656

Re: [asterisk-users] 99 bottles of beer

should be extended to control x10 devices as well... > > "To control the tv in this room, press 1. To control a tv in another room, > press 2. To control the outside lights, press 3. To control the > sprinklers, press 4, ..." Only in America ... ;) Regards, Philipp Kemp

Re: [asterisk-users] SET EXTENSION

... Why mess around with the dialplan when you call a PHP script anyway? Do it in the script. Or pass 1${EXTEN} as an argument if you really need to. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems a

Re: [asterisk-users] Cdr reports

Steve Murphy wrote: > So, the billsec field is usually SMALLER than the duration. Usually? Are there situations when the duration is smaller than billsec? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems

Re: [asterisk-users] How many calls can use the same username

nf: call-limit Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de My pick of the month: rfc 2822 3.6.5 Ges

Re: [asterisk-users] Snom 300 Hints and LIne Buttons

atently not! > > This obviously makes the BLF lamp on my Snom 370 pretty useless as it > only lights up when the Snom 300's got two lines going :-( > > Can anyone point me in the right direction to getting this fixed? Do your peers in sip.conf have call-limit=, e.g. call-limi

Re: [asterisk-users] GotoIf not working with ${EXTEN} for me in 1.4.8

because in Asterisk the quotes are part of the string. But your subject implies that this expression works for you with variables other than ${EXTEN} or Asterisk before 1.4.8? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use

Re: [asterisk-users] [OT] Zaptel 1.2.20 and 1.4.5 released

the first time when I realized that FIrefox automatically tries to use the ftp:// scheme when I enter ftp.digium.com in the address bar. Not sure if I like this kind of "smart" software. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Le

Re: [asterisk-users] Locating Asterisk documentation after installation

MOSBAH ABDELKADER wrote: > After installing Asterisk, i have installed the docs by "make progdocs". > > But i don't know where to locate this documentation. Maybe /usr/src/asterisk-*/doc/api/ ? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwie

Re: [asterisk-users] Macro Overlap

Mojo with Horan & Company, LLC wrote: > set your own mutex using astdb? It may just be atomic enough for you to > get by. "atomic enough" - that's a nice term :-) Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de

Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..

Diego Iastrubni wrote: > On Thursday 09 August 2007 09:40, Philipp Kempgen wrote: >> So if there is no problem with your system just don't enable >> debug mode. > See Tzafrir's reply, that would explain much more. That message did not yet make it to me. The list

Re: [asterisk-users] Monitor doohicky got event Event 160 on channel..

fic issue. ---cut--- So if there is no problem with your system just don't enable debug mode. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http

Re: [asterisk-users] Siemens Openstage & Asterisk ?

ot; and assign a phone number. Grüße, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de My pick of the month: rfc 2822

Re: [asterisk-users] Queue stats

om logger script directly from the dialplan with TrySystem() for example. Another option would be to listen to events from the manager interface (AMI) but that's probably not what you are looking for. Hope that helps. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566

Re: [asterisk-users] Slow list

lity that the list server is trying that address > first. It's the other way round. http://en.wikipedia.org/wiki/MX_record#MX_priority : "The MX record with the smallest preference number has the highest priority and is the first server to be tried." Regards, Phil

Re: [asterisk-users] Slow list

Doug Lytle wrote: > Looks like mail is getting held up between INXS.digium.internal and > lists.digium.com Here's what I get: ---cut--- Received: from lanai.amooma.com ([127.0.0.1]) by localhost (lanai.amooma.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id DYhd6TXxmG6B;

[asterisk-users] Slow list

Since the list was switched over to API-Digital almost every message I get is older than a week. Coincidence? Is anyone else having trouble? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones.

Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy

Timothy Parez wrote: > I have my custom sounds which should be played instead of the default > ones when a user is busy or unavailable: > > "The person at extension XXX is not available right now, please" > > Of course I can simply replace the files, but the problem is my > implementation s

[asterisk-users] Slow list

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeye

Re: [asterisk-users] Needed changes in Asterisk to change the SIP port to 5062

Crazy Boy wrote: > sip.conf: bindaddr=5062 bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) bindport=5062 ; UDP Port to bind to (SIP standard port is 5060) Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems a

Re: [asterisk-users] Centos kernel source

Tzafrir Cohen wrote: > Thanks for the detailed procedure. One update: > > On Sun, Jun 03, 2007 at 03:31:53PM +0200, Philipp Kempgen wrote: > >> Here's what I did: >> >> yum update >> >> reboot >> >> yum install kernel-devel-`uname -r`

Re: [asterisk-users] Centos kernel source

Steve Totaro wrote: >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen >> Sent: Sunday, June 03, 2007 8:22 AM >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Centos kernel source >> >> On Sun, Jun

Re: [asterisk-users] Which KDE editor to edit Asterisk config files ?

C F wrote: > just text filez works for me. So you don't use any editor at all? ;-P Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsf

Re: [asterisk-users] Re: RE: Digital Phones

Hans Witvliet wrote: >> On 5/10/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: >> As I know from AVAYA (I am AVAYA certified) that >> digital phones are connected to digital cards and it >> does not go through ethernet switches at all, digital >> phones should be ind

Re: [asterisk-users] SNOM 360 Rejecting Calls

Hi, Is this a real question? I'm asking because it seems to be a babelfish translation of a post in a german forum http://www.ip-phone-forum.de/archive/index.php/t-99696.html from over a year ago. (is their date wrong?) Dovid B wrote: > Does someone know coincidentally the cause for the error m

Re: [asterisk-users] Manager API Output

Arun Kumar wrote: > Is there any way that I can store my manager API output that is: > My question is that is there any why using that I can get the QueueStatus > and store the result in some text file for further processing. > > > $strHost = "127.0.0.1"; >

Re: [asterisk-users] Large dial plans and variables

Doug Garstang wrote: > I have a large dial plan here with over 3000 lines, and several dozen > macros. As it grew, it became apparent that there was some problems. > > 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, > if that macro calls another macro, and passes argument

Re: [asterisk-users] Large dial plans and variables

Doug Garstang wrote: > I have a large dial plan here with over 3000 lines, and several dozen > macros. As it grew, it became apparent that there was some problems. > > 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, > if that macro calls another macro, and passes argument

Re: [asterisk-users] How many users can be supported simultaneously?

Antonopoulos Angelos wrote: > I have a pc with the following characteristics: > > Pentium IV 2.4Ghz HyperThreading > 512 MB PC3600 Dual DDR RAM > Seagate 80GB SATA HDD > 4-port ethernet 10/100 PCI Card > Netgear MA-311 802.11b Wireless Card > On this machine runs a VPN server, an Apache server

Re: [asterisk-users] Send Variable in Dial

Andres Gomez wrote: > I need send a data to sofphones screen when I use a Dial () . SendText()? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asteri

Re: [asterisk-users] Two devices registrating same extension

Frederico Madeira wrote: > Is it possible to asterisk manage multiple devices registration with > same extension ? > When asterisk receive a call for that extension, it send call to all > devices registered with that extension, and rtp go to first one that > answer the call. Not really but I thin

Re: [asterisk-users] Cisco 7920 sccp

Khaled Chehab wrote: > I am trying to register cisco 7920 to asterisk using sccp since to sip > firmware upgrade to it ,but its ends with failed registration.Can you > please send me a sample for sccp.conf configuring cisco 7902. This is my second hit from Google: http://www.voip-info.org/wik

Re: [asterisk-users] SNOM 360

Him Tim, Tim Koehler wrote: > have tried to switch of RTP Encryption on the snom (in the account/identity > connected to Asterisk). Nice to know you're on the list. :) Grüße, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and

Re: [asterisk-users] tone generation

Jerry Geis wrote: > Does asterisk have a way in the dialplan to generate tones? > Say I want to play a tone 300Hz for 3 seconds. > Can I do that? core show application PlayTones > If not, can I use some system command to generate the wav file > then just have asterisk play it? core show applica

Re: [asterisk-users] Softphone that supports central provisioning?

Tim Panton wrote: > Snom used to have a softphone that emulated one of their hardphones. > I don't know if they still do, or if the emulation extended to the > config managment, > might be worth a dig According to Snom they will stop to maintain their softphone. Too much work they say. So

Re: [asterisk-users] Softphone that supports central provisioning?

Stephen Bosch wrote: > Hi, Tzafrir: > > Tzafrir Cohen wrote: >> Dear Senad, >> >> The setup program for your soft phone can be downloaded from here: >> http://malwareserver.com/malware.exe";>http://LINK >> >> During the setup you will be asked for configuration file. Please use >> attached file.

Re: [asterisk-users] Exten Length

Arun Kumar wrote: > I've configured my exten.conf for few exten. But I'm curious to know how > long can be my exten like (exten => XXX.). Is there any limit for > this or not. For asterisk the answer is in include/channel.h: ---cut--- #define AST_MAX_EXTENSION 80 /*!< Max lengt

Re: [asterisk-users] Asterisk config for Apple IPhone

Stefan Wintermeyer wrote: > Am 21.04.2007 um 19:49 schrieb Crazy Boy: >> I want to buy Apple IPhone mobile. How to configure my "Asterisk" >> server in this mobile? Is this mobile supports VoIP configuration? > > hmmm... lets do some searching: > http://www.google.com/search?q=iPhone+SIP > http

Re: [asterisk-users] Softphone that supports central provisioning?

Gordon Henderson wrote: > On Sat, 21 Apr 2007, Philipp Kempgen wrote: > >> Tim Panton wrote: >>> What's your objection to a softphone in java ? > >> Java is slow and the interface is always ugly and doesn't fit >> into the window manager etc. yo

Re: [asterisk-users] Softphone that supports central provisioning?

Tim Panton wrote: > On 21 Apr 2007, at 13:06, Philipp Kempgen wrote: > >> Tim Panton wrote: >> >>> On 21 Apr 2007, at 03:21, Philipp Kempgen wrote: >>> >>>> Tzafrir Cohen wrote: >>>> >>>>> On Fri, Apr 20, 2007 at 11:48:2

Re: [asterisk-users] Softphone that supports central provisioning?

Dean Collins wrote: > Lol, yep you missed something but do you really want to be taught something > you already think you know? ;) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones.

Re: [asterisk-users] Softphone that supports central provisioning?

Tim Panton wrote: > On 21 Apr 2007, at 03:21, Philipp Kempgen wrote: > >> Tzafrir Cohen wrote: >> >>> On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: >>>> Has anyone found a softphone that supports pulling it's >>>> co

Re: [asterisk-users] Softphone that supports central provisioning?

Tzafrir Cohen wrote: > On Fri, Apr 20, 2007 at 11:48:20AM -0400, James FitzGibbon wrote: >> Has anyone found a softphone that supports pulling it's configuration from a >> central server via TFTP/FTP/HTTP, much like hard desk phones use? > > Why would you want to do that? Because you could provi

Re: [asterisk-users] Problem with queue

Sanjay Rajdev wrote: > Regards, > Sanjay Rajdev > Tha i did because i dont want any call to get disconnected. > Can you let me know what can be the problem doing so. > > > - Original Message ----- > From: "Philipp Kempgen" <[EMAIL PROTECTED]>

Re: [asterisk-users] Problem with queue

Sanjay Rajdev wrote: > I have queue set up in realtime on Asterisk 1.4.2. > > Below is the senario that is happenening :: > I have created a test queue with only one agent. Once I call the test queue > the agents phone rings if the aagent is logged on. everything till here is > fine. > Now if

Re: [asterisk-users] SIP INFO message

Peder @ NetworkOblivion wrote: > I've got a very strange problem and I can't figure it out. I have a > Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I > see callerID name, but it is not getting to * via SIP. I am running * > 1.4.2 and the latest Cisco IOS for my router

Re: [asterisk-users] Execute EAGI script with params fromextensions.conf

Griepentrog Scott wrote: > I don't think you can put arguments to the agi. Try is as: > > > exten => 492,2,eagi,InfMsg > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of equis software > Sent: Wednesday, April 11, 2007 10:54 AM > To: asterisk-users@

Re: *****SPAM***** [asterisk-users] RE: Zaptel 1.4.1 Install Modules CentOS

> Software zur Erkennung von "Spam" auf dem Rechner > > priamus.teamware-gmbh.de > > hat die eingegangene E-mail als mögliche "Spam"-Nachricht identifiziert. > Die ursprüngliche Nachricht wurde an diesen Bericht angehängt, so dass > Sie sie anschauen können (falls es doch eine legitime E-Mail

Re: [asterisk-users] Different devices for asterisk!!!

Rizwan Hisham wrote: > Im trying dial a user according to the device s/he uses. i mean if the user > is using asterisk as a peer, then i have to pass the extension in the dial > application like this: > Dial(SIP/[EMAIL PROTECTED]) ;so that s/he can perform routing according to the > DNID. > > and

Re: [asterisk-users] CRM integration with Asterisk

S. A. Kamran wrote: > I just want to add here that Star Outlook Dialer (Free Edition) has > built in integration (through StarJunction) with SugarCRM as well as > with Salesforce CRM. It is available for free download at > http://www.starutilities.com/staroldialer.htm Can someone tell me how t

Re: [asterisk-users] Configuration assistance needed.

J. Oquendo wrote: > Philipp Kempgen wrote: >> How can people know that your asterisk system is where whois >> says you are? >> >> >> Regards, >> Philipp >> >> > A start would be to get the contact information and actually "CONTACT&

Re: [asterisk-users] Configuration assistance needed.

J. Oquendo wrote: > Salvatore Giudice wrote: >> Where are you located? >> >> -- >> Salvatore Giudice >> [EMAIL PROTECTED] >> >> VoIP Security Training, LLC >> http://VoIPSecurityTraining.com > > > Is it me, or do people not know the value of whois

Re: [asterisk-users] Queue call distribution

Jordan Novak wrote: > I have noticed that asterisk will only try one interface per queue at a > time. Is there any way get get it to dial say three at a time and > connect the first one that it reaches. Do you have autofill=yes in queues.conf? Regards, Philipp -- amooma GmbH - Bachstr. 126

Re: [asterisk-users] Asterisk 1.2.17 and BRIstuff

Dominik Zalewski wrote: > I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is > that current version of BRIstuff is for Asterisk 1.2.14. > > BRIstuff 0.3.0-PRE-1y (* 1.2.14) > > If I'm misunderstanding how to apply patches for 1.2.17? Download Bristuff 0.3.0-PRE-1y-e: ht

Re: [asterisk-users] Pound # key not being handled

Alberto Alonso wrote: > I am trying to use call parking. I have the following > in features.conf > > [general] > parkext => 700 > parkpos => 701-720 > context => parkedcalls > > When I try #700 from my softphone asterisk just passes it > and doesn't interpret it. > > Can someone tell me what I

Re: [asterisk-users] Adding DND to dialplan

Brian McEntire wrote: > One question... are there any places to get extra sound files like > "activated" or "deactivated" or "do not disturb is..." ?? I didn't > find them in the sounds directory after a vanilla install of the > latest stable asterisk 1.4. Maybe the asterisk-sounds tarball has s

Re: [asterisk-users] asterisk and mplayer

Jerry Geis wrote: > Odd question here but if I have asterisk running on PC (and mplayer > installed). > and a video phone calls up the asterisk PC can that video image be > played on mplayer? Afaik Asterisk does not have any support for video streams yet. Regards, Philipp -- amooma GmbH -

Re: [asterisk-users] Play "blank" sound while VM recording?

Charles Ulrich wrote: > I have an Asterisk system deployed at a customer's site. It is connected > to the outside world by a local SIP provider. When someone calls in > through the trunk to leave a voicemail, Asterisk is not sending any RTP > packets back through the trunk after the beep is pla

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

john beaman wrote: > I too was curious about this, so I copied the text into Babel Fish, and this > is the result: > > I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of > my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric > Buzay. > > If this guy

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

Philipp Kempgen wrote: > /kick him ;) I'm not really sure whether "fb" is to blame or if mailman should be in charge of filtering autoresponders. Thus I did not put fb on my black(mail)list - yet. :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - ht

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

Jay Moore wrote: > Ok, I'll bite. This is the 4th message like this I've gotten today. I > don't speak French but it looks like an autoresponder. "I'm away from ... to ... I'm going to respond to your message when I'm back. In urgent cases contact ... or ..." > If so, why is it > replying b

Re: [asterisk-users] RE: Asterisk-Addon-1.4.0 MySQL module

KC wrote: > I still can't figure out why res_config_mysql module not showing up with many > attempt. Anyone have any idea on this? > > checking for mysql_config... /usr/bin/mysql_config > checking for mysql_init in -lmysqlclient... yes > configure: creating ./config.status > config.status: crea

Re: [asterisk-users] Adding DND to dialplan

Bruce Reeves wrote: > exten => *73,1,Answer() > exten => *73,n,Wait(0.5) > exten => *73,n,Set(DB(${CALLERID(number)}/DND)=1) Would prefer Set(DB(${DND/CALLERID(num)})=1) > exten => *73,n,Playback(do-not-disturb) > exten => *73,n,Playback(enabled) > exten => *73,n,Hangup() > > and then > > When

Re: [asterisk-users] Adding DND to dialplan

Brian McEntire wrote: > I've read Asterisk should be able to activate a do not disturb feature > to turn off the ringers on extensions. I checked the wiki and can't > find documentation for how to do it. > > Here's my attempt, added to extensions.conf: > > [dnd-on] > exten => _#78,1,Answer > ext

Re: [asterisk-users] Zaptel 1.4.1 Install Modules CentOS

Chris Blunt wrote: > I have a CentOS server that I am trying to configure Asterisk on 1.4 on. > > Everything seems to go ok, with regards to compiling Zaptel, Libpri, > Asterisk (will be using kernel 2.6 timer and ztdummy) > > Unfortunately I can't insmod / modprobe ztdummy. Did you yum install

Re: [asterisk-users] Sponsored development - Monodirectional audio handling

Edoardo Serra wrote: > The purpose of thisi implementation is to deal with some carriers that > give us the call as ANSWERED when the called party is still ringing. > Our billing software is billing the user (and the carrier is billing us) > even with unsuccessful calls. How about you try a dif

Re: [asterisk-users] [MACRO-SCREEN] and MACRO_RESULT

Andy Hester wrote: > exten => s,n,Set(TIMEOUT(response=10)) Should be exten => s,n,Set(TIMEOUT(response)=10) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://w

Re: [asterisk-users] Announcement: Asterisk Service Provider Edition v1.0 Beta

Olle E Johansson wrote: > Asterisk SPE Nice. ;) Was SpitShare developed by project 0401? Didn't read carefully. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> htt

Re: [asterisk-users] Question on Priorities

--[ UxBoD ]-- wrote: > [inbound-sip] > exten => uxbod(u1),1,Dial(sip/1001,20,t) > exten => uxbod,n,PlayBack(uxbod) > exten => uxbod,n,Hangup() > exten => uxbod,u1+101,PlayBack(uxbod) > exten => uxbod,n,VoiceMail([EMAIL PROTECTED],s) > exten => uxbod,n,Hangup() > > but when I do a extensions reloa

Re: [asterisk-users] Question on Priorities

Yuan LIU wrote: >> From: "Rizwan Hisham" <[EMAIL PROTECTED]> >> Date: Sat, 31 Mar 2007 17:01:51 +0500 >> >> [inbound-sip] >> exten => uxbod,1,Dial(sip/1001,20,jt) >> exten => uxbod,n,Hangup >> >> exten => uxbod,102,PlayBack(uxbod) >> exten => uxbod,103,VoiceMail([EMAIL PROTECTED],s) >> exten => ux

Re: [asterisk-users] Re: asterisk-users Digest, Vol 32, Issue 129

[EMAIL PROTECTED] wrote: > Je suis absent du 2/04/2007 au 11/04/2007. Do not auto-reply to list messages. :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://

Re: [asterisk-users] Indicating agent status on the phone

Alexander Topolanek wrote: > I'm playing around with the queue features, and I'm looking for a > solution to indicate the agent status (logged on / logged off) on the > phone (Grandstream GPX2000 or SNOM 190) > > It would be nice to use one of the busy lamps... Can be done with BriStuff and the

Re: [asterisk-users] Realtime call-limit

Philipp Kempgen wrote: > Peder @ NetworkOblivion wrote: > >> Does anybody know the sql type for the "call-limit" field under sip >> peers? Everything on voip-info is missing that entry. > > Realtime reads/writes everything as a string. But you may as

Re: [asterisk-users] Realtime call-limit

Peder @ NetworkOblivion wrote: > Does anybody know the sql type for the "call-limit" field under sip > peers? Everything on voip-info is missing that entry. Realtime reads/writes everything as a string. But you may as well use a tinyint default null. It's important to name the field "call-limit

Re: [asterisk-users] Multi-Level Queue

Peder @ NetworkOblivion wrote: > I want calls to come into a queue and do a ringall on a number of phones > (let's say 3). So ring them for 20 seconds or so. If there is no > answer, I want it to ring a second set of phones for 20 seconds. If no > answer, then go back to the first set of pho

Re: [asterisk-users] Need help to strip variable

van Leen, Phil wrote: > EG: > With $(SIPURI) I want to keep just the sip number and delete the remainder > '@server.com'. > > Ideally I'd like to use 'SayDigits($([EMAIL PROTECTED])' Maybe Set(without_host=CUT(${SIPURI},@,1)) ? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied

Re: [asterisk-users] CallerID + Name

Rob Schall wrote: > We have the caller id with name option enabled with our provider, > however, our polycom 501 phones will only display the number of the > incoming call. Is there a way to see the callerid name from the cli when > the call is coming in (like a print in the dial plan)? NoOp(Call

Re: [asterisk-users] help - UNSUBSCRIBE

Jerric wrote: > Please remove this email from your mailing list. > > UNSUBSCRIBE ---cut--- List-Unsubscribe: , ---cut--- ---cut--- To UNSUBSCRIBE or update options visit: http://lists.digium.com/mai

Re: [asterisk-users] Dial(Local/[EMAIL PROTECTED])?

Rizwan Hisham wrote: > I dont understand the syntax of the dial application when used like this: > > Dial(Local/[EMAIL PROTECTED]) > > i want to know what is this "Local" doing instead of Tech like SIP, IAX, > H323? SIP/200 would dial a device (the SIP user 200) whereas Local/200 dials the exte

Re: [asterisk-users] Dell poweredge 860 acceptable for office environment ?

Tim Panton wrote: > I really like the Mac Minis as small office servers, quiet, cool, > real UNIX, asterisk works on them. > The only downside is that you can't add PSTN cards. Aren't there USB adapters available? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www

Re: [asterisk-users] Conference server (or how to make a call withmore than 3 u

Jon Schøpzinsky wrote: > Use Snom phones. > We have had around 6 participants, without problems. In theory you should be > able to have around 12 people on a conference on a snom phone. I don't think this is true. The Snoms do not have enough CPU power for 12 people in a conference *on the phon

Re: [asterisk-users] camp on off-line phone

Jeronimo Romero wrote: > It would be cool if you could add some kind of login script capability to > nodes in sip.conf and iax.conf. > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen > Sent: Sunday, March 18,

Re: [asterisk-users] camp on off-line phone

Leif Neland wrote: > When phone A registers, I want phone B to ring, when picked up, it should > call phone A and connect the phones. > > Translated: When GF in Mexico powers up laptop where soft iax-phone > registers automatically, I want to talk to her asap :-) > > How to? I don't really kn

Re: [asterisk-users] How to access Voicemail Password in Asteriskwithout using V

Tzafrir Cohen wrote: > On Sat, Mar 10, 2007 at 05:04:01PM +0100, Philipp Kempgen wrote: >> jamshed zaidi wrote: >> >>> sir, i have configured voicemail already but the problem is i want a >>> variable which holds a value of password just like ${VM_CALLERID} var

Re: [asterisk-users] How to access Voicemail Password in Asteriskwithout using V

jamshed zaidi wrote: > sir, i have configured voicemail already but the problem is i want a > variable which holds a value of password just like ${VM_CALLERID} variable > holds Voicemail Caller ID (Person leaving vm). i want to process password > set by the user through comedian mail. my snerio

Re: [asterisk-users] How to access Voicemail Password in Asterisk without using VoicemailMain????????

jamshed zaidi wrote: > can anyone tell me that how to access voicemail password without using > VoicemailMain is there any variable where asterisk stores it or in > db. voicemail.conf? extension_number => voicemail_password,user_name,user_email_address,user_pager_email_address,user_option(

Re: [asterisk-users] AEL #include file

Thanks for the reply! Steve Murphy wrote: > At Digium, for instance, we keep all our config files under > SVN, and the config > files are just #exec's for svn checkouts. Nice. > Just one little mistake I hadn't pointed out earlier; the > extensions.conf would probably really be extensions.ael !

Re: [asterisk-users] AEL #include file

Steve Murphy wrote: > On Fri, 2007-03-09 at 20:42 +0100, Philipp Kempgen wrote: >> Tzafrir Cohen wrote: >> >>> On Fri, Mar 09, 2007 at 07:49:45PM +0100, Philipp Kempgen wrote: >>>> Hi, >>>> >>>> Does anyone know how to include a file in

<    3   4   5   6   7   8   9   >