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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] 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,${EXTEN},1)
[local]
exten = _7XX,1,Dial(SIP/${EXTEN},30,wtr)
exten = _7XX,n,VoiceMail,u${EXTEN}
exten = _7XX,n,Hangup
exten = _7XX,102,VoiceMail,b${EXTEN}
exten = _7XX,n,Hangup
___
--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] 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 3 of the MSN (local) numbers to ring at the same time. I can't get more than 2 phones to ring at the same time, unless I ring them all by dialing the group, but that's not what I want. ___ --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] 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 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] 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 Goto(s,4) , with n you dont know where you wanna go. regards KAi ___ --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
[asterisk-users] DIAL and automatic/manual co line acces
Hi list, I've got following Problem: i have severel phones on my asterisk. and externel lines connected (POTS sip, does not matter) a externel caller A (CID(num)=0815) calls me ( 4711) . 4711 can be distributet to severel internal extensions for example 23 and 42. 23 is on ZAP/1 and 42 is on ZAP/2 23 has manual co line acces (does not relly matter how this is realized). (i.e. He has to dial a 0 befor he can dial an extenal destination number) 42 has automatic co line acces (no zero in front) if neither 23 or 42 heared the call (out of office. bla) and want to recall the caller by pushing the redial button on his/her phone, the number has to have a leading 0 at extension 23 and none at extension 42. i could alter the CALLERID(num) to 04711 and then do a DIAL(ZAP/1), when there is only one internal destionation, but as i have two internal destionations, DIAL(ZAP/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-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 schrieb: Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? thanks klaus ___ --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 ___ --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] 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... actually it does not, and i have no really running version of this http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro idea at the moment. Maybe you get this on the run... lemme plz know! Regards KAI ___ --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] 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 - installed asterisk 1.2.10 (from ftp) - got the latest from http://beronet.com/downloads/install-misdn-mqueue.tar.gz ___ --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] 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 visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --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?
No, i cannot open a bug on this, cause i dont have a PRI that uses zap. so if there were any questions, you had to answer them. But there is a similar bug, using mISDN. http://bugs.digium.com/view.php?id=7435 and a solution for ME, dont know if it will help you: http://fuhrmannek.de/projects/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 from where you parked the call, should ring after 45 seconds (default) does this work for you? (which asterisk version dou you have?) 1.2.9.1 bristuffed and no it does not seem to work. It seems to mixup src and dst channel: == Parked Zap/4-1 on 701. Will timeout back to extension [from_internalisdn] s, 1 in 300 seconds The call came from another extension and another context. Therefore the callback will fail (and _does_ fail)... Will you file a bug report and give me the bug number? ___ --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 ___ --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?
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 an idea. btw. this is not directly connected to your problem, but: 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 from where you parked the call, should ring after 45 seconds (default) does this work for you? (which asterisk version dou you have?) regards KAI ___ --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] Voicmail Question
is it possible to pick up a call from VoiceMail system? If you mean to grab a call that is currently in the Voicemail application, then no - This is what i wanted to do, exactly. okay, not possible. It's not a favorably (is this the right word in english) answer, but now i know, it's 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:
What about DIAL ( |M(macro-name))
and set the userfield in cdr during execution, ...
Set the userfield to what? That is the entire problem. ${CHANNEL} will give me
something like Zap/10-1. ${BRIDGEPEER} is empty. I would love to see the called
MSN in the port-field something like Zap/10-43 if MSN 43 was called... :-) That
would help enourmously.
Kind regards,
JP
___
--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
___
--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] 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 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?
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 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?
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 called party.
give it a try and let me know, what callerid contains.
___
--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
___
--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
[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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 not the extension dialed. E.G. a Dial(Zap/10/43) will result in a CDR Zap/10-1 which does not help me unfortunatly. Any ideas? Kind regards, JP ___ --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 ___ --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] 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 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] ACD Queues Agents logout
hmm looks nicer than mine:
exten = *2002,1,System(asterisk -rx \
agent logoff Agent/ ${AGENTBYCALLERID_${CALLERID}})
exten = *2002,2,Playback(agent-loggedoff)
exten = *2002,3,Hangup
thx for your suggestion, i think i will integrate your solution
regard
KAI
Anthony Rodgers schrieb:
Hi Kai,
This is what we do:
[agent-login]
exten = s,1,NoOp(${AgentUser})
exten =
s,2,AddQueueMember(${AgentContext}|${AgentChannel}|${AgentPenalty})
exten = s,3,Wait(1)
exten = s,4,Playback(agent-loginok)
exten = s,5,Hangup
exten = s,103,RemoveQueueMember(${AgentContext}|${AgentChannel})
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,
autologoff=time
but how can an agent explicit log off?
regards
Kai
___
--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
___
--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
___
--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] 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. but i dont think at all, this is a solution: if your product is good at all, people would come back and ask, if they want to alter something. if your product is bad... you have to obfuscate things!! regards KAI Marcus Carlson schrieb: Don't really know if this is possible but the way I think it works it should be doable. Have the configfiles encrypted and decrypt when asterisk is starting/reloading and then encrypt again. Marcus Eric Bishop skrev: Anyone know if it possible to create binary/obfuscated/ human unreadable extensions.conf/sip.conf etc.? We would like to deploy a system in an environment where not giving out root is still not enough. We want to hide the contents of these normally plain text files. ___ --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 ___ --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 ___ --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] 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 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] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
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. bristuff makes a vendor check. i dont know the difference between bristuffed and chan_capi-cm (?) kann us tell anyone? How did you finally manage to compile bristuff on Centos 4? Sorry, i didn't try. i used CentOS for a fileserver for a friend of mine. where i had to compile a new kernel but as i asked. can you send error messages? Yours Kai ___ --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] 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 mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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] 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 sources in system. I didn't use bristuff autamated install. wget-ed asterisk, libpri, zaptel and patched them. there is recomended to use make linux26 when making zaptel on 2.6. kernel. bristuff compile.sh don't have linux26 option that linux26 stuff is as far as i know only important to ztdumm.ko, a kernel module which is needed, if you have no Zaptel Cards in your PC and want to use MeetMe Conferencing system. you dont need to tell zaptel wheter you have a 2.6 or 2.4 Kernel, the Makfile discovers this himself. so, no need to worry about 2.4 or 2.6 stuff. Getting kernel sources was a torture for me on Cent-OS 4. maybe somebody can explain how to get them the right way!!! and apply the patches and that. Which Cards do you wanna use in your asterisk (especiallly which ISDN cards, if any) can you post the errormessage of the bristall install script? regards Kai ___ --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] 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:
does /dev/zap exist?
maybe some issues with devfs/udev and dabian?
___
--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] 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: immediate=yes, runs exten = when phone is picked up Cisco and others: Look up PLAR ___ --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] 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 don't hear the start (or the first part) of the
voiceprompt.
It makes no differece if I use the Playback or Background Command.
But it makes a difference if the prompt is played automatically during
entering the queue ore if I playback the file manually by typing 3
for example:
exten = s,7,Background(01_ni_asterisk-b) - missing start part at
beginning of prompt
exten = 3,1,Playback(01_ni_asterisk-b)- sound is ok (whole
soundfile is played)
Here is my extensions_queues.conf
[menu-it]
exten = s,1,Set(LANGUAGE()=de)
exten = s,2,system(/bin/echo ${LANGUAGE} /tmp/LANGUAGE)
include = sipint
exten = s,3,Answer
exten = s,4,SetMusicOnHold(default)
exten = s,5,Set(TIMEOUT(digit)=3)
exten = s,6,Set(TIMEOUT(response)=16)
exten = s,7,Background(01_ni_asterisk-b)
exten = s,8,Background(queue-menu-announcement)
exten = s,9,SetCallerID(IT Queue: 0${CALLERID})
exten = s,10,Queue(queue-it|t)
exten = s,11,Playback(nbdy-avail-to-take-call)
exten = s,12,Voicemail(u1700) ;
exten = s,13,Set(LANGUAGE()=en)
exten = #,1,Hangup
exten = t,1,Hangup
exten = o,1,Goto(menu-office,s,1)
exten = i,1,Playback(invalid)
exten = i,2,WaitExten
exten = 3,1,Playback(01_ni_asterisk-b)
exten = 4,1,Playback(02_ni_asterisk-b)
Thanks for any help.
morel
___
--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
___
--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
[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
; ${ARG2} - Device(s) to ring
;
---BeginMessage---
Hie,
I trie to use a simply call forward, found on this mailing list (:-),
when i'm not near my phone:
i creat a global set:
olscell=123456789 ; my cell phone number
A macro for forwarding the call:
[macro-cell_user]
exten = s,1,Playback(Call_Transfer)
exten = s,2,Flash()
exten = s,3,SendDTMF(${ARG1})
exten = s,4,Hangup()
I put in m incoming context:
exten = 0470022762,1,Dial(IAX2/300,20,tr)
exten = 0470022762,2,Macro(cell_user,${olscell})
But, when the call is being, the phone is hangup!
What do i do in macro for forward the call??
Best regards,
--
Olivier Saulnier
STEGANUX
1er étage DIAMECANS
BEL AIR
03410 St-Victor
T: 04.70.02.27.62
F: 04.70.09.97.41
http://www.steganux.com
___
--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
---End Message---
___
--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] 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 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] 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 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] 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 ever going to buy an VoIP-Phone. any suggestions for this situation? (i.e. which devices do you prefer) Kai ___ --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] 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 = *32,1, disable cfwd [mailbox] exten = *41,1, enable mailbox exten = *42,1, disable mailbox Thanks again. But I want to ask what is the usage of include if it is a continue-until-matched type of contruct. ___ --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] 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 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] 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.
___
--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] BLINDTRANSFER
http://svn.digium.com/view/asterisk/trunk/UPGRADE.txt?view=markup
* Builtin transfer functionality sets the variable ${TRANSFERERNAME} in the new
created channel. This variables holds the channel name of the transferer.
I'm not sure if channel name is enough for me, but i made the the cid_num
available by myself
in the same manner as trunk does it with TRANSFERERNAME.
(not exactle the same line in res_features.c but i altered
this today...)
so don't drive yourself crazy. i got it working allready
regards
Kai
C F schrieb:
I don't realy get it, what are you trying to accomplish? 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
therefor b has to know who which callerid --transfered-- him.
___
--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
___
--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
___
--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
[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 callerid --transfered-- him. is there a VARIABLE or something else, where i can look up WHO transfered b? thx Kai ___ --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] 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% sure, but i think the card or the asterisk, or the isdn stack, will not recognize any new lines added during a running asterisk session. Giorgio Incantalupo schrieb: Hi, I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything seems ok, asterisk gives no error (nothing inside logs) but the 4 led on the back of the card (which is NOT connected to an ISDN line) are red and flashingwhat does it mean? Is it not properly working or it means the card is not connected to any ISDN line? The card handbook says the card has red led but not their meaning. TIA Giorgio Incantalupo ___ --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 ___ --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] 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 CPU. a friend of mine mentioned, that if there are many SIP participiants in the conference the delay will even be greater. (one guy told a joke, and laughing was delayed from 10 to 30 seconds) regards Kai thanks we are planing to have around 50-60 users in 1 room. ___ --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
[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 Dial(mISDN/9-u73, mISDN/2| |t) in new stack Jun 18 18:55:36 WARNING[8461]: chan_misdn.c:4452 chan_misdn_log: misdn_call: No Extension given! -- Couldn't call 2 == Everyone is busy/congested at this time (0:0/0/0) == Auto fallthrough, channel 'mISDN/9-u73' status is 'CHANUNAVAIL' how to avoid this sinc Kai ___ --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
[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, adam hears a beep, and knows there is an other caller on line. - bob and adam stop seaking - adam hangs up (i think this happens: - eve sends her callerid to adam and then asterisk segfaults ) when i disable usecallerid=yes ; show CLID on incoming calls in zapata.conf, asterisk does not crash i'm not sure if it matters clid signalling, but i live in germany. okay, any ideas but disabeling CLID ? any further information needed? sinc Kai *CLI -- Starting simple switch on 'Zap/8-1' -- Executing Dial(Zap/8-1, Zap/3||tT) in new stack -- Called 3 -- Zap/3-1 is ringing -- Zap/3-1 is ringing -- Zap/3-1 is ringing Jun 18 21:49:46 WARNING[11353]: chan_zap.c:1419 zt_train_ec: Unable to request echo training on channel 3 -- Zap/3-1 answered Zap/8-1 Jun 18 21:49:46 WARNING[11353]: chan_zap.c:1419 zt_train_ec: Unable to request echo training on channel 8 -- Attempting native bridge of Zap/8-1 and Zap/3-1 -- Hungup 'Zap/3-1' == Spawn extension (analog-8, s, 1) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' -- Starting simple switch on 'Zap/8-1' Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 18 (Ring Begin)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:00 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 17 (Polarity Reversal)... Jun 18 21:50:01 NOTICE[11361]: chan_zap.c:6062 ss_thread: Got event 2 (Ring/Answered)... -- Starting simple switch on 'Zap/3-1' -- Hungup 'Zap/3-1' Ouch ... error while writing audio data: : Broken pipe Warning, flexibel rate not heavily tested! Segmentation fault ___ --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] 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 //not sure, if this is really neaded Hi I've problem with incoming call quality to GSM gateway connected to beronet card (BN8S0), - [ GSM Gateway ] --- [ BN8S0 ] asterisk ___ --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] 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-init.conf: card=1,0x8,pcm_slave,ignore_pcm_frameclock //important! option=9,master_clock // 9 for port 9 pcm=1,1 //not sure, if this is really neaded Intresting I'm going to try this today . I thinking also about 'ulaw' option to 'card=' . My channelbank is T1 and this will eliminate transcoding from isdn to T1.i hmm, my S0 cards are connected over a pcm bus ( the BN8S0 provides this, ). I don't think the pcm stuff will solve your problem, but hey, give it a try kai ___ --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
[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/${ARG1}sip-out)
exten = s,n,Hangup()
when i use my analog telephone, everything is okay:
- Starting simple switch on 'Zap/3-1'
-- Executing Macro(Zap/3-1, anrufextern-sip|9199125) in new stack
-- Executing SetCallerID(Zap/3-1, SIP-ID) in new stack
-- Executing Dial(Zap/3-1, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip-out-0fe9 is ringing
-- SIP/sip-out-0fe9 is ringing
== Spawn extension (macro-anrufextern-sip, s, 2) exited non-zero on
'Zap/3-1' in macro 'anrufextern-sip'
== Spawn extension (macro-anrufextern-sip, s, 2) exited non-zero on
'Zap/3-1'
-- Hungup 'Zap/3-1'
but when i dial from my isdn phone, it dials as soon as it gets the
first digit of the phone number
and does not wait for the 199125
-- Executing Macro(mISDN/2-u12, anrufextern-sip|9) in new stack
-- Executing SetCallerID(mISDN/2-u12, SIP-ID) in new stack
-- Executing Dial(mISDN/2-u12, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip-out-5d40 is circuit-busy
any ideas?
are there any switches in the misdn.conf providing this?
using :
misdn 0.3.1-rc11
asterisk 1.2.7
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem dialing out thru sip - using isdn on internal
got following hint from c.richter from beronet support team
exten = _8.,1,waitfordigits(4000)
exten = _8.,n,Macro(anrufextern-sip,${EXTEN:1})
exten = _9.,1,waitfordigits(4000)
exten = _9.,n,Macro(anrufextern-analog,${EXTEN:1})
now it gets all digits
___
--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
[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 google, but i searched for over an hour, but can't find anything. maybe i use the wrong search words, anny suggestions? thx Kay ___ --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: 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: http://lists.digium.com/mailman/listinfo/asterisk-users
