similar to: how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?

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 =>

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

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'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

-----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

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

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

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

Hi, Just started using an asterisk-based PBX with Polycom IP501 phones. Am Fairly satisfied and am starting to get into FTP setup of the phones. Have figured out most things except for how button remapping works. In sip.cfg, I have this entry: <keys key.IP_500.31.function.prim="DoNotDisturb"></keys> This works as expected but if I try to change the remapping to any

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Hi, I am trying to use the attended transfer. So I put this in my feature.conf: [general] [featuremap] atxfer => *0 blindxfer => #0 I completly restart asterik, and not just make a RELOAD. But during a call, when I press # it runs a blind transfer and if I press * I am disconnected. I am using the CVS version of * get as explain here

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

> 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

Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the "answer" field is always empty, "billsec" field is 0 and "disposition" field is

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 am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on

>Kind of OT here, but just out of curiosity, how do you email them? Do >you have an actual address, or do you just use the form on their web >site? I've sent a bunch of requests via that form, and even though it >says I should receive a response, I never have. I've tried going >through a couple of resellers, too, and haven't gotten anything. I'm >hoping you

Hi, I'm looking for a multi-line sip/ip phone that can answer multiple incoming paths. IE a secretary sits at a front desk and can answer multiple incoming lines/DID's. Is there something like this? Thanks, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041202/5b685d69/attachment.htm

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]