RE: [Asterisk-Users] Web based Asterisk management tool
Why not share with the community? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Chandler Sent: Wednesday, February 09, 2005 9:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Web based Asterisk management tool Gary, contact me off-list. I have developed a GUI Windows based tool that will allow management of configuration files if you are running RealTime. It supports sip,iax,extensions,voicemail currently. It will also display CDR's and the various schema's used by the Asterisk box. Tom Chandler [EMAIL PROTECTED] - Original Message - From: "dean collins" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, February 09, 2005 10:47 AM Subject: RE: [Asterisk-Users] Web based Asterisk management tool > You need to go back and reread. > > It is just pretty much an asterisk configuration tool (ok some minor things in the backend but it's the best out there). > > AMP is available for free download but they make their money by offering support. > > [EMAIL PROTECTED] is just an automated way of installing AMP and FOP and Web Meetme. > > If you really have a need to support thousands of extensions as you suggest then you should really go back and learn how to program asterisk with a database yourself from scratch. > > > > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brett, Gary > Sent: Wednesday, February 09, 2005 11:17 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > Thanks for the responses, isn't [EMAIL PROTECTED] an pre-configured distribution > for small systems?? I am looking for an open source web management tool to > use on any size asterisk server (even ones that are already up and running) > the user base could be anything between small and large with many external > lines, > > Ive looked at AMP, is it free ? and are there any alternatives or is AMP the > only open source web management tool ? > > -Original Message- > From: dean collins [mailto:[EMAIL PROTECTED] > Sent: 09 February 2005 15:05 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > That would be the AMP database, I don't know. > > Ping the amp list and find out. > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa > Sent: Wednesday, February 09, 2005 9:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > How big can be [EMAIL PROTECTED] user data base? Can it handle 1000s of users ? > > Regards. > > Daniel. > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of dean collins > Sent: mercredi 9 février 2005 15:42 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Web based Asterisk management tool > > Hi Gary, do a search for [EMAIL PROTECTED] the iso is available for download > at sourceforge and does exactly what you are looking for. > > > Cheers, > Dean > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Brett, > Gary > Sent: Wednesday, February 09, 2005 8:01 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Web based Asterisk management tool > > > Hi there > > I am new to Asterisk and am looking for a web based management tool, for > managers to manage hunt groups, extensions etc and for user to have > access > to there own phone features. I have seen there are a number of > commercial > tools available for this, but I presume there are some freeware options > too > > I noticed one that I like at http://www.thirdlane.com/screenshots.htm > but I > am assuming this is just a freeware product that has been re-badged so > to > speak. > > If any body can give me some suggestions that would be great > > Regards > Gary > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > __
[Asterisk-Users] back to operator
Is there a way to have a caller go back to the operator once they are in the voicemail directory or are they stuck? IE I call in and don't' know the extension but go to the company directory and can't find who I want, how do I get back to the operator? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR---if you know your parties extension you may dial it now
I have searched and I have my IVR working when it has to fork off to another application but how do I get it to allow callers to dial the extension directly instead of going though the directory? [mainmenu] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ;SAI menu - 1 for tech support, 2 for directory, 3 for echo test exten => s,5,Background(custom/sai-welcome) exten => s,6,Background(custom/sai-choose) ; Leave Voicemail ;exten => 1,1,VoicemailMain2() ;exten => 1,2,Hangup ; company directory exten => 2,1,Directory(local) ; Echo Test exten => 3,1,Playback(demo-echotest) exten => 3,2,Echo exten => 3,3,Playback(demo-echodone) exten => 3,4,Goto(mainmenu,s,6) I have extensions that start with 3's, 9's, 1's and 6's ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bellster - IAX-based interchange -- lets you callanywhere for free
I get a 500 Internal Server Error when I try to register. :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Guy Sent: Friday, January 21, 2005 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Bellster - IAX-based interchange -- lets you callanywhere for free Fellow Asterisk Users, The original Free World Dialup vision was to facilitate world wide calling using each others phone lines. The service grew significantly, but is pretty much limited to Internet calling and contacting other ITSPs. Now, Jeff Pulver has created Bellster(tm) - Half Napster/Half Party Line - to fully realize the original vision. We've just finished our testing and it is now open for your use. We'd love to hear your feedback. It is Asterisk(tm) based and you connect using IAX. Visit www.bellster.net for information and to sign up. Best Regards, Ed Guy, pulver.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Xfering a call
Well that didn't workI now get this error
Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569/5", "b") in new
stackJan 12 16:56:21 WARNING[4989]: app_voicemail.c:1539 leave_voicemail: No
entry in voicemail config file for ''
-- Timeout on IAX2/[EMAIL PROTECTED]:4569/5
== CDR updated on IAX2/[EMAIL PROTECTED]:4569/5
-- Executing Goto("IAX2/[EMAIL PROTECTED]:4569/5", "#|1") in new
stack
-- Goto (home,#,1)
-- Executing Playback("IAX2/[EMAIL PROTECTED]:4569/5", "sai-thanks")
in new stack
Jan 12 16:56:31 WARNING[4989]: file.c:475 ast_openstream: File sai-thanks
does not exist in any format
Jan 12 16:56:31 WARNING[4989]: file.c:779 ast_streamfile: Unable to open
sai-thanks (format ulaw): No such file or directory
Jan 12 16:56:31 WARNING[4989]: app_playback.c:83 playback_exec:
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569/5 for sai-thanks
-- Executing Hangup("IAX2/[EMAIL PROTECTED]:4569/5", "") in new stack
== Spawn extension (home, #, 2) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569/5'
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/5'
This user does have an entry in the voicemail.conf file..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, January 12, 2005 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Xfering a call
> I'm having an issue when I transfer a call to another SIP extension it
sees
> that the sip phone is not there and goes to voicemail but in my case it
> transfers to the main voicemail instead of the users voicemail.
>
> Here is what my SIP extensions look like in the extension.conf file
>
> exten => 3957,1,Dial(${Theresa},20,Tt)
> exten => 3957,2,VoicemailMain2(u${TheresaVM})
> exten => 3957,3,Hangup
> exten => 3957,102,VoicemailMain2(b${TheresaVM})
> exten => 3957,103,Hangup
Change the above from VoicemailMain2 to Voicemail and it will work
as expected.
The 3,Hangup isn't required... remove it. 103 isn't actually needed
either.
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[Asterisk-Users] Xfering a call
I'm having an issue when I transfer a call to another SIP extension it sees
that the sip phone is not there and goes to voicemail but in my case it
transfers to the main voicemail instead of the users voicemail.
Here is what my SIP extensions look like in the extension.conf file
exten => 3957,1,Dial(${Theresa},20,Tt)
exten => 3957,2,VoicemailMain2(u${TheresaVM})
exten => 3957,3,Hangup
exten => 3957,102,VoicemailMain2(b${TheresaVM})
exten => 3957,103,Hangup
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[Asterisk-Users] OT question
Can someone help me answer this question? Where would you most likely find a file with the line "+::"? What does it do? I have been racking my brain a buddy of mine is testing me and I don't want him to win. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FWD to IVR
I'm trying to setup my FWD# so that when it reaches my * my IVR plays.
Currently this works by me having it go to a dummy SIP/# as shown here in my
extensions.conf
exten => ${FWDNUMBER},1,SetMusicOnHold(default)
exten => ${FWDNUMBER},2,Dial(SIP/},10,tr)
exten => ${FWDNUMBER},3,Wait(2)
exten => ${FWDNUMBER},4,Answer
exten => ${FWDNUMBER},5,Wait(1)
exten => ${FWDNUMBER},6,Goto(mainmenu,s,1)
Is this the correct way? Is there a better way on doing this?
Thanks
Mike
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[Asterisk-Users] Can't dial out on X100P
I've been beating my head against the wall all night on this. I have gone through about 200 google searches trying to figure this out and am at a loss. I did do the instructions outlined in the readme.linux26 for FC2. Below is the error I get when I try to dail out. Dec 7 22:28:49 WARNING[2202]: channel.c:1920 ast_request: No channel type registered for '' Dec 7 22:28:49 NOTICE[2202]: app_dial.c:803 dial_exec: Unable to create channel of type '' (cause 66) == Everyone is busy/congested at this time Inbound work fine, and transfer's to my SIP phone with out a problem. I just can't dial out. Ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. Zapata.conf [channels] language=en ;context=inbound-analog context=default signalling=fxs_ks usercallerid=asreceived echocancel=400 echocancelwhenbridged=yes group=1 channel => 1 Zaptel.conf fxsks=1 loadzone=us defaultzone=us ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco 7940 help
Does anyone have any simple documentation on converting a 7940 to SIP and making it function with *? I have been beating my head on a wall on this and have gone no where. Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk gui?
I agree it was nice once it was configured but it did blow up my install of *. Since it was on a test server I went and nuked it all, but will just keep my eye on it as it evolves over time. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Oliver Stone Sent: Wednesday, November 24, 2004 7:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk gui? I had to give up after attempt to install AMP.. It's got very nice user interface, that is, AFTER you have sucessfully installed it.. I see it's got great potential, but current release is very difficult to install, even with the "newbie guide". If you have fewer than 10 extensions to configure, it's probably not worth your effort to going thru all the trouble to install AMP., hoping you didn't screw up anything during the install process. . Just wondering how difficult it would be for AMP devs to develop a install wizard or a batch file that can automatically execute the install and download necessary dependencies... until then, I guess I'll be continuing to manually config my asteisk files On Mon, 22 Nov 2004 23:12:00 -0700, Brian <[EMAIL PROTECTED]> wrote: > > Perhaps rather than a GUI we should be wanting an IDE (as in Integrated > > Development Environment, not Intelligent Drive Electronics . . . bloody > > overlapping acronyms . . . but I digress . . . ). > > > > Even some basic syntax highlighting would improve the readability of > > extensions.conf immensely. Anyone know how to make THAT work in vim? > > I've hacked one together for UltraEdit that works reasonably well, but > > that's a Windows editor. > > Jim, > Several months ago I was working on a VIM Asterisk syntax highlighting file, > but stopped working on it due to lack of interest. I might try to add some > stuff to it again if I have some spare time later this week. > > What I had done can be found at http://snurl.com/asterisk_syntax_vim . > Contributions are welcome; just hit the edit button on the wiki :) > > -Brian > > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
