similar to: User interface issues (was voicemail menu structure)

Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on

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

If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the

Thanks. Is there a way I can log into the Merlin Magix to determine that? How else do I tell? ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Germann Sent: Monday, February 05, 2007 8:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users]

Hey All, I'll be configuring an asterisk box to be the voicemail server to an old Merlin system which had an octel 100 voicemail server that is now dying. My question is simple: do I need to stick an FXO card in the asterisk box? My logic is that if the Merlin Magix system is actually generating electrical current, then I would need to have an fxo card. Is this correct?

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

Hi list, Is this supposed to be a joke? It doesn't sound very professional. comedian n 1: a professional performer who tells jokes and performs comical acts Moby Thesaurus words for "comedian": banana, buffoon, burlesquer, card, caricaturist, choreographer, clown, comedienne, comic, cutup, dramatist, dramatizer, dramaturge, droll, epigrammatist,

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 anyone else finding in the new audio files that the longer space at the beginning and end of the files tends to be extremely irritating? An excellent example is when going into voicemail and Allison says how many messages you have, the space between the files is annoyingly long: you have ...... four ...... old ...... messages ..and.. first ...... message ...... received ..... July .....

Hello, everybody: I'm developing an application using Asterisk and a TDM-400 card. I understand the concept of the difference between GSM and WAV files when using Asterisk, but I'm not happy with the sound quality with the GSM compression. It's merely *acceptable* for a telephone call, but for anything else, it leaves something to be desired. Case in point -- if you compare the

Hello: I've been wrestling with an integration issue for which there also seems to be one piece missing. I'm hoping someone on this list can help. Due to a complete lack of cooperation from our current voice provider we are in need of an alternative way to access our Octel 350 voicemail system from our SIP (SER) proxy. I can redirect and relay calls to numerous destinations via

Hi All, I'm having a frustrating time exporting a GPFS Independent Writer AFM fileset through Samba. Native GPFS directories exported through Samba seem to work properly, but when creating an export which points to an AFM IW fileset, I get "Access Denied" errors when trying to create files from an SMB client and even more unusual "Failed to enumerate objects in the container:

Hello all. I am a little (allot) lost on my next hurdle in getting an asterisk system built. I would like to get my asterisk servers configured exclusively from database. I have read through the wiki on this subject but once again I find that there is a certain level of knowledge that is assumed. As of now I know nothing about databases in general and specifically MYSQL. I do not know

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

Steve Blair writes > I can redirect and relay calls to numerous destinations via >SER but because the Octel needs an SMDI interface for mailbox >identification I am stuck, none of the solutions thus far support >SMDI-SIP munging. > > I just started thinking about the possibility of using Asterisk >with a few FXS cards to provide the gateway between SIP and >the Octel.

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 -

Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file. The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes. For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets

Hello: We offer several options for accessing Comedian mail via Asterisk and IP phones. We'd like to add one more. We would like to be able to save a message on the Asterisk server in such a way that it can be retrieved using the telephone set and Comedian mail as it is today and/or via an IMAP client. If accessed via am IMAP client we would like to use an "externotify script" to

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer