similar to: asterisk server as a voicemail server forlegacy PBX -- FXO or FXS???

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?

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits

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

I'm running Asterisk 1.2.8 with Polycom ip501's xten softphones The only problem I'm experiencing is the following: I can't seem to get blind transfers to work with my Polycom 501 phones Either through the feature code or the soft keys. Feature code blind transfers: I set up a feature map in features.conf like this: blindxfer => # This works for all my

Management just approved purchase of a Digium T100P and a T1 card for our Merlin Legend Switch. I will appreciate comments from anyone performing this installation before: - Which T1 card did you use in the Merlin Legend? - Did you require any special interface? (CSU/DSU, etc) - Any items to watch during installation. Also we will need to remove one of our old analog Trunk cards to accomodate

We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF & intercom right out of the box. They can also be centrally managed and provisioned. They also sound great

Hello, I am looking to replace my company's Avaya Merlin Magix system with an Asterisk based PBX when our current lease is up. I had a meeting with upper management yesterday, and they would like some assurance that other companies are running Asterisk with success. We are a relatively small company, with about 70 total extensions. I would be purchasing a Dell Poweredge 2850

I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to

Does anyone know if sip phones from any of the major IP phone vendors support 802.1x authentication? Any feedback would be greatly appreciated. Thanks in advance. ====================== Jeronimo Romero EUS Networks Email: jromero at euscorp.com Cell: 917-332-7238 Office: 212-624-5943 Web: www.euscorp.com ====================== -------------- next part -------------- An HTML

Running Asterisk 1.2.9. I just installed a TE110P card and configured zaptel.conf & zapata.conf. The config files look right to me but I'm getting the following error when trying to start asterisk: Asterisk died with code 1. Automatically restarting Asterisk. Does anyone have any idea what is wrong with this configuration?? Thanks in advance!!! Here's my config files: zaptel.conf

Is there any technical difference between a T1 cable and a cat5e patch cable as far as using them with Digium T1/E1 cards? Can PRI circuits terminating at a smart jack connect successfully to Digium cards using straight through CAT5e cables? If so, are they using all of the pins in the cable? Thanks in advance

Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

I was wondering if people have experienced issues with Cisco 7960G and Asterisk. Any feedback on people's experience deploying this phone in production environments would be appreciated. Thanks in advance

<snip> > Paul Crick wrote: > > Brad Bergman wrote: > > my thinking is that Comedian Mail is its own thing with > > its own interface and users who have become accustomed to > > it, and it needs refinement before it needs an Octel emulator > > I guess it's each to their own. Maybe * could come with a default > Comedian Mail configuration file then have

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.

Hi there, I'm resending this to the list as I didn't get an answer from sfrench.. > I hope I'm right here. I just followed the "Ask the developer" link on > the CIFS homepage... > > I have to mount a little share from a Windows server, which I did with > smbmount in the past. Now the server has been upgraded to Win2003, and > I'm facing the

Hi List, I have a question about uninstalling and installing R on linux, which I am new to. > sessionInfo() R version 2.10.1 (2009-12-14) x86_64-unknown-linux-gnu locale: [1] LC_CTYPE=en_US.UTF-8 LC_NUMERIC=C [3] LC_TIME=en_US.UTF-8 LC_COLLATE=en_US.UTF-8 [5] LC_MONETARY=C LC_MESSAGES=en_US.UTF-8 [7] LC_PAPER=en_US.UTF-8 LC_NAME=C [9] LC_ADDRESS=C

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

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