similar to: How much to use Dialogic?

Aloha, I have a Cisco Gateway the is functioning as my SIP Gateway to the PSTN. I have an Asterisk box, RC2 with a for port FXS card providing dialtone for a Norstar Key System. I have it working so when you press a line key on the Norstar you get dial tone from the Asterisk box. The user has to dial '9' then they can dial there number which is sent to the Cisco GW via SIP and the call

Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML & can get input from the keypad & softkeys. Aloha, Matt

Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Aloha, Are there any hard phones that support ILBC? I see lots of soft phones but not Cisco, Polycom etc. Aloha, Matt

Aloha, Does anyone have a good source for Polycom SoundPoint? IP 600/500/300 phones? Everyone sells Cisco 79XX. Aloha, Matt

Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt

Aloha, Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to. The real issue is the 'end call' button becomes 'forward' when the call ends....therefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up

I'm attempting to use an asterisk box with a Digium TDM01B as voicemail for an existing Meridian/Norstar PBX with an ATA-2 adapter. We're having problems where hangup is not always (but sometimes) detected. It's not detected probably 70% of the time or more. (The users transfer callers to an extension--caller then has to navigate a menu to get to the appropriate user). After

Aloha, Does anyone have any screen shots of the Asterisk admin GUI? I couldn't find any links in the archives or the asterisk web site. -Matt

Aloha, Does anyone know how to have a hot keypad on a Cisco 7960? It allows you to dial on-hook without press the SPEAKER button. Very handy once you get used to it! -Matt

Has anyone been succesful pushing a VLAN setting to a Polycom phone via DHCP? I can push the boot server via option 66 but that is about it. I have set it for 'fixed' and tried many different option numbers with a couple differnet DHCP servers. SIP firmware 1.3.4 or 1.4.1 doesn't make a difference. Aloha, Matt

>From the Wiki: 'There are aready a few bugs in 1.5.2 but more fixes some good new features' Anyone know where the bugs are being listed? I am working through a few issues: 1. When rebooting, the phone will pause for exactly 180 seconds with the screen reading 'updating initial configuration'. I know it is 180 seconds because when I tail the FTP server log the entries are

Hi there! We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of

Why hasn't someone found 50 people who sound alike, put them in sound studios and record the 10,000 most commonly used words. You would all differnent forms of the 1,000 most words, i.e. leading, trailing, question etc. You can synthesize the other 0.05% when you run into them. With hard drives so big, processors so fast and EXT3 that can handle 30,000+ files in a single directory that

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

To All; My current issues is a 5 second delay for call that is being transferred from the Norstar units to the Asterisk servers VIA a PRI. Is their anything that can be done to speed up the transfer on the Norstar. Below is my current phone config. < Norstar1 >----PRI----< Asterisk-1 >----IP-WAN----< Asterisk-2 >---PRI---< Norstar2> The Norstars are MICS 0x32 4.1

Does anyone know which signalling protocol works when connecting the output of a NT8B90AL Norstar ATA2 analog terminal adapter to an FXO card? (fxsks, fxsgs, fxsls?) I'm trying to add an Asterisk branch to my Norstar PBX as outlined at http://www.voip-info.org/wiki-Asterisk+Nortel. - Mike

Hi all, I've seen a couple of posts recently from people who are doing something with Asterisk and a T100P and a Nortel PBX. However it's not clear exactly what they are doing. Does anyone know if it's possible to use a T100P or T400P as a DSX-1 interface to connect to a T1/PRI CO module in a Nortel PBX? We have a Nortel Norstar Modular ICS PBX and I'd like to plug its CO

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls