similar to: Find a person

I have Asterisk version 1.0-RC1 running on Debian Woody. I have 1 analog phone working, 2 inbound lines working, X-Lite is working. The problem that I am having is with Cisco 7960 with SIP version 7.2 software. I can make outbound calls and they work fine, I even get a notice that I have voice mail on the phone and it seems to register properly but I can seem to dial to the phone.

I am using Cisco 7960 phones and have had a request to have the receptionist phone ring on multiple phones just in case she is not around. Call pickup is the theory here but the issue is that not all the people that need to hear the ring would here the receptionist phone ring so I think I need to have a second line appearance on the phones in question so that line will ring. Can this be done

I've seen a lot of traffic on the list recently about which phones can do supervised transfers and which cannot, and I have to admit that I'm a bit puzzled. Our existing PBX, which is software based, handles the transfer functions for our call center -- the agents never touch their phone, and instead use software. We can plug any old phone into it, and it'll work just the same. So

I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten => 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten => 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper

I have been trying to get multiple lines on the 7960 to work for several days. i have read all the posts I can find and have run multiple "sip debug" and have gotten no place on this. Here are the relevant section of the config files: sip.conf [scott] type=friend host=dynamic username=scott secret=scott context=default mailbox=6101 callerid=Scott Henderson [scott1] type=friend

Scott, Right now I am using the Netphone KE1020A. ~Dan Message: 11 Date: Sat, 08 Jan 2005 13:34:29 -0900 From: Scott Henderson <scott@finite-tech.com> Subject: Re: [Asterisk-Users] Re: Connecting Sip phone to asterisk. To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <41E05FF5.30302@finite-tech.com> Content-Type:

I have set up an auto attendant and all is working but I am bothered by a long pause when switching between menus. This pause is between 5 and 7 seconds and is quite annoying. Is there anyway to address this. One other thing I find interesting is that when I move from the main menu to the sub menu the delay is there but when I move from the sub menu to the main menu the delay is not there.

That's the least ambiguous subject I could muster. I'm relatively new to Asterisk and while I'm certain there is a way to do this, I'm unsure how. My question is this: How do I take an incoming call, put the person on hold, and in the background (i.e. while they are on hold) begin trying other phone numbers until someone answers? Example: Jane calls the Network Operations line.

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

I am trying to find an ATA that will provice IAX2 and g729. I have not had much luck, I am hoping someone here might have some ideas. -- Scott Henderson ============================================================================ Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com

I have just finished setting up a new asterisk system which is basically the same as our first system. We are using 7960 phones and I used the phone config files the first installation with appropriate changes. The problem is that on the new system I get no message lights, I can't figure this out. One thing I do notice is that when I monitor the sip debug on the second system the sip

I need to be able to dial a long distance provider that uses an access number and an auth code. I would like to be able to program this so that the user can dial 8 and then the long distance number, asterisk will hopefully do everything in the middle. The sequence to accessing the provider is on my traditional phone speed dial as: * Dial local access number * Wait 5 seconds * Dial the auth

hello Scott. On 08/23/2016 04:40 AM, Scott Robbins wrote: > On Mon, Aug 22, 2016 at 10:47:06PM -0500, geo.inbox.ignored wrote: >> >> greetings. >> >> in an attempt to display correct fonts in firefox instead of squares >> with binary values, i installed wrong fonts and made things worse. >> >> how do i find out what fonts are, as i did not think to

I have encountered a frustrating problem with the meetme rooms and calls entering the system on the Digium analog cards. The typical scenario is: Callers on SIP phones, X-lite, Eyebeam, Cisco 7960, IAXy Callers entering the system from the PSTN via the digium Analog card (TDM400P) In the meetme room the SIP connections can all hear each other loud and clear. The PSTN people can hear

When using Xten's Eyebeam software I am noticing that I get a blocked port 5067 on my firewall. The source port obviously varies but the 6057 seems to be consistent. I have done some looking and can find any reference to what may be happening here. I am guessing I need to modify some packet filters but I would like to make sure I understand this so I can open the right port ranges. --

Short course: Statistical Learning and Data Mining Trevor Hastie and Robert Tibshirani, Stanford University Sheraton Hotel Palo Alto, CA Feb 26-27, 2004 This two-day course gives a detailed overview of statistical models for data mining, inference and prediction. With the rapid developments in internet technology, genomics and other high-tech industries, we rely increasingly more on data

Short course: Statistical Learning and Data Mining Trevor Hastie and Robert Tibshirani, Stanford University Georgetown University Conference Center Washington DC September 20-21, 2004 This two-day course gives a detailed overview of statistical models for data mining, inference and prediction. With the rapid developments in internet technology, genomics and other high-tech industries, we

Short course: Statistical Learning and Data Mining Trevor Hastie and Robert Tibshirani, Stanford University Sheraton Hotel, Palo Alto, California February 24 & 25, 2005 This two-day course gives a detailed overview of statistical models for data mining, inference and prediction. With the rapid developments in internet technology, genomics and other high-tech industries, we rely

Hello, I want to discuss my idea on ranking in IR system which I think can be good extension to Xapian. If I am not too late to discuss it then please consider it. I first give you brief background of me, I am a Masters student working on my thesis in the Information Retrieval. I today only got a mail from one of the professor from Europe whom i am going to join for Ph.D about GSoC and more

I am looking for a SIP Soft Video phone, which I can use with Asterisk. If you have one installed (regardless if free or purchased) please tell me which one, the settings in Asterisk and your experience with it. bye Ronald