similar to: configuring trunks

Hello All, I am getting my first PRI installed in a couple of weeks and I wanted to ask for a little advice. I have a single span Digium card I will be using for the install. Id there a benefit to which protocol I use? When asked, I told them to set it up as NI2. The PRI is through MCI and will be used for local and long distance with DIDs and features like CallerID, etc. Any advice would be

Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000?

Hi list, anybody any example how to use it? I did not find any hint in the wiki nor in the mailinglist archive :-(. I want to use one button showing my agents the actual state (logged in or logged off) Thank you Gerd

Hello, I am still researching my dive into Asterisk at my workplace, and I was wondering about how switchboard activities are handled.. Right now, a call comes into our switchboard, and the operator forwards them to the appropriate line, thus freeing up the primary number and allowing more calls in. Everyone on campus has a direct-dial line as it is right now. I want to eliminate most of

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber:

Dear all! W2k, R 2.5.1 I am working with an ongoing malting barley variety evaluation within Sweden. The structure is 25 cultivars tested each year at four sites, in field trials with three replicates and 'lattice' structure (the replicates are divided into five sub blocks in a structured way). As we are normally keeping around 15 varieties from each year to the next, and take in 10 new

Anybody know where to find Thailand DIDs that can ring in to my * in the USA on SIP? Oh, and a good price, too! ;) Chris Coulthurst chris@shuksan.com

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

Hi; Has anyone successfully relayed an Icecast stream using Shoutcast? I realize that this is backwards! However, several organizations that want to use our streams use Shoutcast, I really don't want to setup shoutcast and Icecast, but unless I can relay an Icecast stream with Shoutcast, I'm going to have to. Thanks, Fred -------------- next part -------------- An HTML attachment was

Hi, I have a matrix with column names starting with a character in [0-9]. After some matrix operations (e.g. copy to another matrix), R seems to add a character 'X' in front of the column name. Is this a normal default behaviour of R? Why has it got this behaviour? Can it be changed? What would be the side effect? Thank you. Regards, Gilbert [[alternative HTML version deleted]]

hi guys: We have just rented a server and setup a BBS for asterisk discussions at http://bbs.us.xgforce.com feel free to join. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050718/07587c97/attachment.htm

Hi, I have a question conercing wondershaper. I''m using the Clarkconnect linux distribution for my linux router and I tried to use wondershaper. On start up of wshaper, there are no errors or any other problems but I''m not sure if it''s running correctly. Only one qdisc / one class is used and when I start an uplink ftp transfer, my ping time is growing up to 1700ms - I

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

The situation is that I want to have certain extensions that can only call nationwide and others that can make international calls (with a different provider). I suppose, you could also see this as a scenario that could involve multiple internal extensions that get matched to specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Hello everyone, I'm trying to turn up a SIP trunk with a Cisco UCM (Unified Communications Manager/Call Manager). It's currently configured for 3rd party call control (3pcc). The INVITEs show up without an SDP... Neither the Cisco admin nor myself can find any documentation on how to disable this feature (3pcc). Does anyone happen to know how to disable 3pcc on Cisco Unified

I am trying to turn up DID trunks with our local phone company but do not know the correct format of extensions.conf to do this.

Hi, I'm trying to configure 2 parallel sip trunks between 2 boxes. However I seem to have the problem that when making a call from Box 2 to Box 1, it sometimes says authentication failed because it is using the username of the other trunk. Here's my configuration: Box 1: [dp-dp2] type=peer username=dp-dp2 secret=mysecret qualify=yes host=box.2.ip.address context=from-internal [e911-dp2]

Hey all, First off, I'm new to cygwin, rsync, and actually even ssh. I've used ssh clients many times, but server side I'm a bit of a noob. Anyway, here's my problem... I set up a brand spankin new Windows XP box with only two apps installed: copSSH and rsync. I need to tie down all security as tight as possible. From what I've seen/read, it seems to me the only port I need