Re: [asterisk-users] User authentication

go here http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate and read ;) Siqhamo Sifo schrieb: How does one configure user authentication on asterisk . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] Playtones

what about this? show app ringing? [incoming] exten = s,1,Answer exten = s,n,ResponseTimeout(5) exten = s,n,Playback(mymessage,skip) exten = s,n,Background(mymessage2) exten = s,n,Background(silence/3) exten = _7XX,1,Ringing exten = _7XX,2,Goto(local,${EXTEN},1) [local] exten =

Re: [asterisk-users] mISDN versus ZapHFC with BRIstuff

is it a single s0 card? how do you ring the 3 phones? no problems with the installation of mISDN so far. it is as easy as on Bristuff regards KAI Henrik Woffinden schrieb: Hi Sorry... I haven't been specific enough... I have several ISDN phones on my inside NT mode ISDN card, and I wan't

Re: [asterisk-users] WAIT FOR DIGIT not working

prints print really to stdout?, flushed the output? $target = ; print WAIT FOR DIGIT 5000\n; $target .= STDIN; print WAIT FOR DIGIT 5000\n; $target .= STDIN; print WAIT FOR DIGIT 5000\n; $target .= STDIN; ___ --Bandwidth and Colocation

Re: [asterisk-users] callback without agi

exten = 333,n,Authenticate(1234) . . exten = 333,n+101,NoOp(Is this ok??) Or i have to explicitly enumerate the priority? ... i'm searching for doc about this. as far as i know Auth( ) does not jump to n+101 if you dont use Auth..(123,j) enumrations are easier if you use somthing like

[asterisk-users] DIAL and automatic/manual co line acces

/1ZAP/2) both would get the 0 for manual co line access. but 42 does not need a leading 0. any other suggestions than rewriting app_DIAL? Thanks for your answers, says Kai Ober ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

Re: [asterisk-users] misdn-init.conf card parameter for a monoBRI

what about a subscription on the Misdn-asterisk@lists.beronet.com mailing list! http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk Regards Kai ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

gcc -v : gcc version 4.1.0 no problems using latest stuff from beronet/downloads/misdn_queue.stuff suse standard kernel regards KAI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Giorgio Incantalupo schrieb: Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. Did you change them? Now everything works fine? regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] transform bridged call into a conference

you're searching for 3pty... which TECH do ya use? zap/misdn/bristuff/sip/ or do ya look for a generel solution? zap does this by itself! there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. Klaus Darilion

Re: [asterisk-users] transform bridged call into a conference

zap does this by itself! how? threewaycalling=yes in zapata.conf there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. ok - how does it work? with app_chanredirect? this was used to run in astersik 1.09...

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Giorgio Incantalupo schrieb: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y so the only i can tell is: - my kernel is 2.6.16.13-4-default (meaning suse 10.1 default) - installed latest zap and libpri packages from asterisk-org -

Re: [asterisk-users] Annoying Bristuff

Does anyone have any other tips. use mISDN ;) or are you bound to bristuff because you need speciall features of this? KAi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] where/when to set__TRANSFER_CONTEXT ?

Hi there, i want to use another context, when i do a atxfer, but i dont know when/where to set that magic variable. in the dialplan, any examples? Regards, Kai Ober ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

[asterisk-users] Has anybody a usefull example for the DIAL-option G(context|exten|prio)

HI lIst, i'm a little confused about the G option of dial. which sense hast it to send calle and caller to an context/extension and dont bridge the calls, is ther a way to bridge the two parties??? Has anybody a usefull example for this option? Looking forward to your answers KAI

Re: [asterisk-users] cmd DIAL - Who picked up the call?

/asterisk/download/res_features-misdn-bugfix.diff good luck KAI Koopmann, Jan-Peter schrieb: On Tuesday, August 01, 2006 9:36 AM Kai Ober wrote: when you park a call (asterisk feature defautl keys: #700 ...) at your isdn phone and you forgot to catch the call on another phone, the phone

Re: [asterisk-users] cmd DIAL - Who picked up the call?

I have had a similar problem a few days ago, when i did a blindtransfer i wanted to know which extension the transferer had. i added a variable my self: pbx_builtin_setvar_helper(chan, BLINDTRANSFERER, transferee-cid.cid_num); i see that this is not what YOU need, but maybe it helps to get

Re: [asterisk-users] Voicmail Question

not possible. thx anyway Kai Ober ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] cmd DIAL - Who picked up the call?

(How do you get to the dial command, can you send the extension for this?) the idea is to to use $EXTEN.call a macro with $EXETN as an argument ... i'm not used to ZAP PRI stuff, nor do i own such a card, so i cant't test what to do. On Friday, July 28, 2006 3:12 PM Kai Ober wrote

Re: [asterisk-users] SRTP

Khaled Chehab schrieb: Is SRTP available in asterisk? Or how to implement it ? am using trixbox you asked this question before, and you got answers, read your mail, or stay away from this list! ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] cmd DIAL - Who picked up the call?

What about ${CALLERID} ??? dont know if it is available during all, and if it's the ID form calling, or called party. give it a try and let me know, what callerid contains. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] cmd DIAL - Who picked up the call?

Okay, i think this wont work. i do some magic whith callerid , when i am the calling party, but it makes no sense, that callerid is my side when i'm called, sorry for this one :) What about ${CALLERID} ??? dont know if it is available during all, and if it's the ID form calling, or

[asterisk-users] Voicmail Question

Hi list, is it possible to pick up a call from VoiceMail system? Didn't find nothing on voip-info.org Thanks for your answers KAI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] cmd DIAL - Who picked up the call?

What about DIAL ( |M(macro-name)) and set the userfield in cdr during execution, ... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Hi, if I do Dial(SIP/peer1/numberZap/g1/Number) can I somehow figure out who exactly picked up the call? In the cdrs dstchannel I can see the channel but

Re: [asterisk-users] ACD Queues Agents logout

I didn't want to send the Agent thru the whoule AgentCallbackLogin rutine just to _log off_. This does not make really sense to me. thank for your answer anyway. Kai Here is what I do... Exten=777,1,AgentCallbackLogin() ___ --Bandwidth and

Re: [asterisk-users] ACD Queues Agents logout

}) exten = s,104,Wait(1) exten = s,105,Playback(agent-loggedoff) exten = s,106,Hangup A. On Jul 20, 2006, at 6:26 AM, Kai Ober wrote: Okay, I think i have missed something: When i use AgentCallbackLogin*(||*007) the agent is logged in, fine. But how do i log OUT. okay there is a timout

Re: [asterisk-users] Binary/unreadable configuration files?

show dialplan or other commands from cli renders this unnecessary. the only way to make those things unreadable, IMHO is an sophisticated,komplex dialplan/extension.conf which is unreadable at all. or an other way may me using as much agi as you can, and an binary exe file which is encrypted.

Re: [asterisk-users] Binary/unreadable configuration files?

so, whats the idea of open source? files. The configuration of a given set is your IP... Most people don't just give that stuff away. okay... dont feed me, the troll, i will stop answering this thread. regards KAI ___ --Bandwidth and

Re: [asterisk-users] Binary/unreadable configuration files?

I'm not sure, but can asterisk-BE do something like that? regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

Cosmin Prund schrieb: I'm using a HFC-S ISDN card and a TDM400P zaptel card with 3xFXO and 1xFXS modules. Okay, getting a HFC-S card, a single s0 card, working with bristuff should be no problem at all. be aware, if you want to use 4s0 or 8s0 Cards other then from junghans bith bristuffed.

Re: [asterisk-users] all call forward

http://www.cyber-cottage.co.uk/wiki/index.php/Call_forward Is it possible to change the value of ${EXTEN}? Or does it have any better way to implement to the all call forward feature? ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] ACD Queues Agents logout

Okay, I think i have missed something: When i use AgentCallbackLogin*(||*007) the agent is logged in, fine. But how do i log OUT. okay there is a timout, autologoff=time but how can an agent explicit log off? regards Kai ___ --Bandwidth and

Re: [asterisk-users] Bad luck installing bristuff on multiple Linux'es. Any one got a good-luck story I can repeat?

Filip Drągowski schrieb: First question: Do You have kernel sources ? this is required for #make-ing zaptel i have installed asterisk-1.2.4 with bristuff-0.3.0-PRE-1k, libpri-1.2.2 and zaptel-1.2.3 OS was Debian 3.1 with kernel 2.6.16.2, i compiled new kernel myself so there was kernel

Re: [asterisk-users] Ztdummy

What I find interesting is that timing will work. However I don't feel comfortable letting the client use the system if this can affect him in anyway. Thanks. Do you have any Zaptel card in the box? GotoIf($[${ANSWER}=YES]?s-yes|1:s-no|1) s-yes: you dont need ztdummy s-no:

Re: [Asterisk-Users] Question setting up a bat phone extension.

Has somebody done that with a Grandstream GXP-2000 or a BudgetTone 100/101 ? Has somebody even a list which SIP phones have this funtion? Regards Kai It's called hotline or Private Line Auto Ringdown (PLAR). SIP: It's a function of the phone, look for hotline in phone docs Zap:

Re: [asterisk-users] don't hear start/begin of voiceprompts

Okay, i will be one of the 100 answering this question. what about a wait (2) before the background()? That should manage your problem. Mein Name schrieb: Hi all, I just want to setup new voiceprompts for serveral queues in our asterisk pbx (Version 1.2.41.2.4) The Problem is, that I

[Asterisk-Users] Forward call

go here http://www.voip-info.org/wiki/view/Asterisk+call+forwarding and look this *sterisk 1.2* [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN} here as well ; ${ARG2} - Device(s) to ring ; ---BeginMessage---

Re: [asterisk-users] IVR DTMF

At /var/lib/asterisk/agi-bin/dtmfivr.sh for example After that what should I do read this book? http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 this webpage http://www.voip-info.org/ regards ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Question setting up a bat phone extension.

Eric ManxPower Wieling schrieb: I don't know about Grandstream devices since they are banned from our network. Banned? I didn't try any other devices, but whats wrong with the Grandstreams?? wondering Kai ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Question setting up a bat phone extension.

Eric ManxPower Wieling schrieb: Grandstream seems unable to produce stable firmware. They have tried for *YEARS* and still people have to try many different versions of the firmware to find one that actually works in their environment. okay, i see, thx :) i will try to remember, if i'm

Re: [asterisk-users] priority problem

I use include in an other way than you do. i use different extensions, not the same in each includet context, maybe that makes more sense (to you) [apps] include = emergency include = cfwd include = mailbox [emergency] exten = 911,1,do stuff here [cfwd] exten = *31,1, enable cfwd exten =

Re: [asterisk-users] AGI tutorials

Rizwan Hisham schrieb: Anybody who knows a good source of AGI tutorials on the net? plz share try one of the mirrors and then the pages on AGI, http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 have Phun Kai ___ --Bandwidth and

Re: [Asterisk-Users] BLINDTRANSFER

thx, but where is the CALLERID in ${BLINDTRANSFER}, ther is just cheannel info. looking for this tech/port|channel/transfererid , not this tech/port|channel http://www.voip-info.org/wiki/view/BLINDTRANSFER therefor b has to know who which callerid --transfered-- him.

Re: [Asterisk-Users] BLINDTRANSFER

? that the CID should be that of who? On 7/3/06, Kai Ober [EMAIL PROTECTED] wrote: thx, but where is the CALLERID in ${BLINDTRANSFER}, ther is just cheannel info. looking for this tech/port|channel/transfererid , not this tech/port|channel http://www.voip-info.org/wiki/view/BLINDTRANSFER

[Asterisk-Users] BLINDTRANSFER

Hi List, i'm fiddling around with a blindtransfers. (and 3PTY) a calls b a transfers b to c (blindtransfer) (c is not a party but a makro which puts b into a MeetMe conference) the conference should be dynamically created. and named after the callerid of a therefor b has to know who which

Re: [Asterisk-Users] beronet BNS40 led blinking: not working or not connected?

Have you startet the asterisk allready? When i boot my machine, and dont start the astersik, the LED's keep flashing all day. (even when lines are connected) and even if /etc/init/misdn_init has been startet TIP: First connect all Lines/Phones to the card, then start asterisk. not 100%

Re: [Asterisk-Users] Meetme max users

Today i put 10 users in a Meetme on a 700MHz machine. but the result did not satisfy me. I had all 10 Phones in front of me, cause i'm testing my asterisk. so i could speak on one phone and listen on any other. i had a delay of 1 sec of my spoken word(s) so i think, that you should use a BIG

[Asterisk-Users] isdn and PARK

Hi outa there, when i park a call which i picked up at an isdn line , the call never comes back, -- Stopped music on hold on mISDN/9-u73 -- Added extension 'mISDN/2' priority 1 to park-dial == Timeout for mISDN/9-u73 parked on 701. Returning to park-dial,mISDN/2,1 -- Executing

[Asterisk-Users] CALLERID problems asterisk segfaults

Hi all, i use asterisk 1.2.7 and i have a problem with callerid. i use sangoma a200 cards. one fxo one fxs module i have these scenario - bob calls adam, where bob calls into my asterisk and adam picks up from my asterisk - bob and adam are speaking to each other - meanwhile eve calls adam,

Re: [Asterisk-Users] sound quality problem on mISDN

Have you only one BN-Card? or more? i have two cards, had compareable problems. PCM was the magic word ... from my misdn-init.conf: card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! option=9,master_clock // 9 for port 9 pcm=1,1

Re: [Asterisk-Users] sound quality problem on mISDN

Piotr Chytla schrieb: On Wed, Jun 14, 2006 at 10:04:04AM +0200, Kai Ober wrote: Have you only one BN-Card? or more? I've one BN8S0 card and one TE110P - T1 connection to Rhino channelbank. i have two cards, had compareable problems. PCM was the magic word ... from my misdn

[Asterisk-Users] problem dialing out thru sip - using isdn on internal

hi, i've a wirded problem, i try to dial out, using this dialplan [default] exten = _*7.,1,Macro(anrufextern-sip,${EXTEN:2}) [macro-anrufextern-sip] exten = s,1,SetCallerID(SIP-ID) exten = s,n,Dial(SIP/${ARG1}sip-out) exten = s,n,Hangup() when i use my analog telephone, everything is okay: -

Re: [Asterisk-Users] problem dialing out thru sip - using isdn on internal

got following hint from c.richter from beronet support team exten = _8.,1,waitfordigits(4000) exten = _8.,n,Macro(anrufextern-sip,${EXTEN:1}) exten = _9.,1,waitfordigits(4000) exten = _9.,n,Macro(anrufextern-analog,${EXTEN:1}) now it gets all digits

[Asterisk-Users] Free/Open pci telco card

Hi List, While I was surfing the net last week, I found a link for open source pci telco cards. I'm not sure if it were isdn or analog related. The layout an all the stuff was free downloadable, so that you can build your own cards. Does anybody have the link? Yes, I know there is

Re: AW: [Asterisk-Users] Free/Open pci telco card

http://www.zapatatelephony.org/ Yes, indeed. THX very much, i would have searched forever ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: