similar to: voicemail advanced options problem with mysql datbase

hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail => odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer("SIP/xx.xx.xx.xxx-0a02e1c0", "") in new stack -- Executing Set("SIP/xx.xx.xxx-0a02e1c0", "foo=102") in new stack -- Executing

Hello, am attempting again to resolve an issue with multi-tenancy and the forwarding to VMs between mailboxes. If in a multi-tenancy environment one uses custom contexts ie. [a1-ext1](a1) mailbox=101 at a1 and the associated voicemail.conf entry: [a1] 101 => 1234,My User 1,addr1 at email.com,,tz=eastern|imapuser=addr1 at email.com|imapfolder=Inbox 102 => 1234,My User 2,addr2 at

Hello all, I am struggling with sending voicemail as an attachement in Email. When i have given the email like someone at gmail.com it is delivering to my gamil account perfectly(of course to spam folder). But when i given the email like someone at mycompanymail.com it is not delivering to my company email account.. What should i do ? Actually my company is using a third party email server..

So before I start I should be upfront and disclose that I''m using external_nodes, so in keeping with that I''m trying as hard as possible to keep all the actually ''specific'' data in the node config files (currently just yaml)and have nothing specific in modules especially. So I have this definition: (credit goes to DavidS for the inspiration) define git::repo (

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

Hello users, Actually i wanted to implement Text-To-Speech engine from cepstral voice using swift application i tried the documentation of doing this and i was unsuccessful at doing this work with asterisk can anybody please help me out finding the solution to installation thanks in advacnce srinivas Antarvedi -------------- next part -------------- An HTML attachment was scrubbed... URL:

> > As a result, I just go back to put all users in [default] in > voicemail.conf. > > Am I missing anything? >> What do those contexts mean in your setup (beside being arbitrary >> groups)? I just want to group the mailboxes by say department rather than putting them all under [default]. So, I could use CLI to "voicemail show users for sales" or

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

>Date: Mon, 4 Aug 1997 10:05:04 +0200 >From: marc.waknine@espace.aerospatiale.fr >To: samba@arvidsjaur.anu.edu.au (Non Receipt Notification Requested) >Subject: printer problem . URGENT help form SAMBA team !! >Message-ID: <9708040805.AA06185@musun95009.espace.aerospatiale.fr> > >on the SAMBA server i declare some printers >everything is ok in win3.11 because all

Hello, We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of voicemail you can press 3 for advanced options, 5 to leave a message and enter an extension to leave a voicemail. This feature worked fine under 1.4. Now under 1.6.1 all the prompts are the same but when you enter the extension it reads back the extension (or says the recorded name if present) then goes straight

Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(XXX at VMContext) so how

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the SkypeUserB's details so that i can control who can forward the calls to my asterisk box. Thanks in

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

Hi all. I usally make backups of databases mysql. I make buckups of all datbase for example: mysqldump -u user -ppassword name_db > backups.sql also I make backups just its schema for example mysqldump -u user -ppassword name_db --no-data > backups.sql but now I need a backups incrementals, because the size of DB is very big (500 mb) How to make this? regards -- *Atte.

hi, I''ve already set up hieraki2 (database.yml: configured, schema.mysql.sql: imported). I noticed that there is no user (table "users" is empty) and I can''t login. So what should I do to use hieraki2? :) The title is "Root | Demo" - Why "demo"? I tried to create account called "root" using "signup" form. I''ve got

On Tue, Oct 20, 2015 at 01:31:46AM +0200, ?ngel Gonz?lez wrote: > On 16/10/15 12:46, hubert depesz lubaczewski wrote: > >On Thu, Oct 15, 2015 at 04:15:03PM -0400, Daniel Kahn Gillmor wrote: > >>> if the intermediary machine (the "jumphost") is jumphost.example, and > >>> you are trying to reach bar.example.com (which is behind the firewall), >

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Hello all, I am testing the Dial application with the fall through priorities for different cases what i want is the flow after failure of the Dial application which simulates response codes like 1)404 -- Not found 2)480 --Temporarily Unavailable 3)486 --User busy i did manipulate the priority flow like the following for the case 2 and 3 ... exten => _XX,1,Dial(SIP/extension) exten =>

before_save :strip_strings validates_uniqueness_of :username validates_presence_of :username private def strip_strings self.username = username.gsub!(" ","") end example: username = " gissmoh-/upiLifMog2ELgA04lAiVw@public.gmane.org " becomes gissmoh-/upiLifMog2ELgA04lAiVw@public.gmane.org and is stored in the