similar to: 0 out of voicemail to different secretaries

Does anyone know of a way to specify what extension is dialed when 0 is pressed in the voicemail system. I have a situation where there is more than one secretary and they want the 0 to redirect to the appropriate secretary for the two groups of people. So an example would be: 555-1234 -> voicemail -> Secretary 1 555-1235 -> voicemail -> Secretary 2 Any help would be greatly

There has been discussions about the voicemail menus and some of us would like to see an overall plan for the voicemail menus. There are 3 primary ways of arranging the menus. First is a tree structure, second is a random access structure and the third would be a hybrid of the two. (Comedian mail is currently a hybrid.) As was pointed out by Brad Bergman, the ideal would be to have it

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, some users are to read other users mailboxes (secretary-boss relationship), in order to filter out minor issues, SPAM, appointment requests etc. pp. The "boss" wants to see/access his/her mailbox the usual way, no fancy stuff. Furthermore, this is a N:N relationship, meaning one user needs to read one set of user's

We are currently migrating from a Nortel pbx to Asterisk and we have been able to convert most of the functions that people are used to but there is one I have no clear idea how to do. The scenario is: Boss calls secretary from outside the office to get connected to another outside destination. The secretary dials the destination and then trasfers call to the boss. When boss finishes

Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. I've tried the following, but it doesn't work: exten =>

Hello, we are running a Asterisk (1.2) installation with about 80 snom phones (300,320,360). Now have the demand for a special manager - assistant setup for a few extensions. Since Shared Line Appearance is not available in 1.2 I?m wondering how to realize this... What we need is that the manager can decide whether he wants to get calls or not. If not he must have the possibility to redirect

We have had a few different times when a user has forwarded their phone to himself. This has overloaded the communications to our operator panel (FOP). One user should not be able to effect the whole phone system! Is there a way that the number of times that a call can be forwarded could be limited like to 10 or even 100? Then even if a user does something stupid like forwarding their calls to

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

Hi, I have been using qmail-ldap for quite some time and now moved to postfix/dovecot. One feature that I miss is that provided by qmail-secretary. qmail-secretary basically is a mail list manager with following features: 1 no limit, just explode to all members 2 members only, as the name says; only members are allowed (based on envelope sender, so not very secure, everybody can fake

I've mostly got our dovecot+postfix+SOGo+openldap open source groupware replacement working the way I want it to; we're replacing GroupWise in our organization and I'm thrilled to be doing it. I'm supporting about 1,000 active staff users (and another 6,000 student accounts). I've got e-mail and calendar sharing working, and it does what it says it will do, but it is (go

Hello, we got a SNOM 360 here and this gota TRANSFER button. With this i can transfer a call from my phone another one. But when i push this Button and transfer the call to another phone, i get kicked out. Now, every secretary first asks the chief if he is available or not - how can i implement this feature thx for any ideas !

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

> d = data.frame(gender=rep(c('f','m'), 5), pos=rep(c('worker', 'manager', 'speaker', 'sales', 'investor'), 2), lot1=rnorm(10), lot2=rnorm(10)) > d gender pos lot1 lot2 1 f worker 1.1035316 0.8710510 2 m manager -0.4824027 -0.2595865 3 f speaker 0.8933589 -0.5966119 4 m sales

1. RE: Start recording automatically when xferring to anextension? (Don Pobanz) Message: 1 Date: Fri, 25 May 2007 11:54:33 -0500 From: "Don Pobanz" <dpobanz@hastingsutilities.com> Subject: RE: [asterisk-users] Start recording automatically when xferring to anextension? To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Dear list, I have a problem to install the GRASS_0.1-4.tar.gz on RedHat7.1 Linux. The gmake5 is missing. But I am not quite sure what it is? Is it a gnu version of MAKE? I could not locate it on my system and I do not know what is really meant. Any experience? I get the following message: ---------------------------------------------------- bash-2.04$ R CMD INSTALL -l

Hi. I am replacing a legacy Alcatel PBX for asterisk and now i'm facing a request i can't answer yet: I need to replicate the "Call Pickup" function from Alcatel, mostly when used by a secretary picking up a call from his manager. The pickup function, as i'm already using her is not enough, i need the pickedup number to be shown in the secretary phone. Is this doable, using

Hi! I have to Set up an asterisk Server with a Diva Server PRI E1-30M. Capi, asterisk, etc. everythink works. my problem is the handling of the MSN's. say, we have the block (without area-code..) 4321-0 to 4321-4999 between this numbers (including em) every MSN is possible. do I have to add all MSN's i need (several hundrets) to the capi.conf? then the routing to SIP-Phones shall be