similar to: Asterisk 1.2 is getting closer - please help

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.7 http://astguiclient.sf.net/ The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is

Thank you very much for all feedback on Asterisk Sunday News! This is the last issue for June. This week I'll go on holiday and will be back with more news in early July. My kids are getting summer leave this week and we'll be visiting the south of England for a while. Another part of Europe that still use their own currency. If you think there's an European standard, you're

I am wondering if there are any tricks getting the Cisco ATAs to do "distinctive rings" via the ALERT_INFO variable? I have seen some contradictory information in the Wiki, and I tried the example there. I then sniffed the connection between the server and the ATA and didn't see the header sent like it is "supposed" to be. If someone out there has a handle on this and

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Hello, I found, that in CVS Head, in chan_sip.c, there's some support of asterisk. I have one question -- how does it map extensions to sip user names? When my client "subscribes" to other extensions' presence, they see all users online, but it may be because of voicemail fallback. Is there a way to map extension to some sip user's presence? Any ideas are welcome.

Is there a way of increasing the delay before asterisk picks up the incoming PSTN call? I'm using a tdm400p with fxo card. It seems to pick up the inbound call immediately. I want to delay detecting the call by about 10 secs if poss. Done some searching but couldn't find anything relevant. Cheers, Taff!!! ___________________________________________________________ Yahoo!

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username

Astricon 2005 will take place in Anaheim, California October 12-14 2005. Astricon is the Asterisk conference, arranged by IPsando LLC in cooperation with Digium. We are now looking for speakers. The conference will be bigger than last year, so we are looking for more speakers in the conference part of Astricon. In addition to the tutorials, we're adding a three-track conference day with a

We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara

Hi, I'm taking a straw-poll to see who out there is planning on going to AstriCon. I would like to hear from both new members of the community and gurus. What kinds of things would you like to see at an Asterisk Conference? What topics are good BOF (Birds Of a Feather - informal discussion group) fodder? What parts of Asterisk require the most attention? FYI - AstriCon is October 12 - 14

Friends, I have updated my Asterisk 1.2 presentation with the latest information. It is still available in the same place as before: http://www.astricon.net/asterisk1-2/ Please continue to test the beta of Asterisk 1.2, available at ftp.digium.com. We need all the feedback we can get. If you are a developer and have some time for community work, please check in with the bug tracker and help us

Friends, Please send all offers for help *off list* to us at info@astricon.net. Do not disturb the list with offers of your services, please. I repeat: Only the Developer's Meeting will be considered for broadcast at this time. In order to enjoy the conference, you will simply have to be there. It's an IRL experience - meeting all the other Asterisk user's from around the globe,

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes

Hello Asterisk users around the world! Recently, I have been working with pretty large Asterisk installations. 300 servers running Asterisk and Kamailio (OpenSER). Replacing large Nortel systems with just a few tiny boxes and other interesting solutions. Testing has been a large part of these projects. How much can we put into one Asterisk box? Calls per euro invested matters. So far,

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be

We're now opening up registrations for the Astricon tutorials again. We've been able to move to new conference rooms within the same hotel. Register on line at http://www.astricon.net We're sorry for the inconvienience our recent closing of the tutorials may have caused you. You are welcome to contact us at info@astricon.net to change your reservation to include the tutorials. There

Sunday news is today published on a monday. Yesterday was fourth of july, and I used that as an excuse for being off line yesterday. (Sweden's national day is June 6th - and it's not yet a public holiday, btw). Most of my Asterisk time lately have been used for producing the registration site for Astricon and tracking down speakers that haven't sent in their material for the conference

How do you configure an outbound proxy for Asterisk? If the sip call is not local I want everything to go to a designated sip proxy. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/bdd57a91/attachment.htm