search for: asteriskdocs

I made myself look a little silly recently in a talk regarding asteriskdocs.org. I didn?t realize the 4th ed. of the Definitive Guide was apparently actually out (http://shop.oreilly.com/product/0636920025894.do), because I went by asteriskdocs.org?s claim that it was being worked on in OFPS (now retired, apparently.) Is the 4th ed. available to read online like the 3rd...

If one of the authors is listening: http://www.asteriskdocs.org/html/apas02.html lists usereqphone 2 times. One of the entries should really be useragent. And the example for usereqphone is wrong. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new on...

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I installed Asterisk 11 via the following command *> svn co http://svn.asterisk.org/svn/asterisk/branches/11* (as written in asteriskdocs.org http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html ) But it seems that I have a development version instead of a stable release: *> core show version* *Asterisk SVN-branch-11-r375559 built by user @ user-MS-6580 on a i686 running Linux on 2012-11-01 13:...

...versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. We have done loopback tests with the Digium card with a loop plug in it. We're a bit stumped...

...degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's topics: ------------------- * Looking beyond Asterisk 1.0/1.1 - what's up? * Asterisk on SlashDot * Asterisk on the O'Reilly Open Source Convention * Writing community documentation - asteriskdocs.org * Application of the week: The MeetMe conference system * Alternatives to the standard MeetMe *** Looking beyond Asterisk 1.0/1.1 - what's up? ------------------------------------------------- At Astricon in September we'll arrange a developer's meeting. This is the first time w...

A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040728/3306e606/attachment.htm -------------- next part -------------- ? Hi, I'm dialling 1234 in the softphone or Grandstream Phone and without disconnecting the phone i want to dial "10" after dialling "1234". Is it possible to do? Regards shan

...is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the source as well as on the following web page: http://dev.asteriskdocs.org If you will be attending Spring VON in San Jose, be sure to come by the Asterisk Pavilion and say hello! A number of the Asterisk developers, including myself, will be there talking to people about Asterisk. Thanks, Russell Bryant "drumkilla" -----BEGIN PGP SIGNATURE----- Version:...

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4 Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I've tried every combination and I cant get it to work. I'm assuming it must work otherwise it wouldn't have been listed but fo...

I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir

...aren't seeing each other at all. Could the side with the high > > LBO be confusing the other side somehow? > > Shouldn't be... you did use a T1-crossover cable to cross-connect the > two Asterisk boxes, right? I've got a little T1 cross-connect diagram > at http://www.asteriskdocs.org/cables/ if you need a reference. > Yes, it's a T1 cross over and this problem was resolved after plugging into a working Digium card on the other end. Brandon B. --0015174be84cb5e5900464293445 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable &lt...

I am trying to find where the authoritative sql definitions for Asterisk Realtime Architecture ARA are located. I have found many locations but each and everyone seems to be different. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example Files included with the distribution: asterisk-11.9.0/contrib/realtime/mysql/iaxfriends.sql:CREATE TABLE `iaxfriends` ( asterisk-11.9.0/contrib/realtime/mysql/meetme.s...

....com] On Behalf Of Scott Griepentrog Sent: Thursday, August 27, 2015 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Local channels: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html This explains adding members to queues, although it doesn't specifically provide an example of using local channels in a queue: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html Basica...

Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Johnathan Falk Network Administrator Clinton Community Schools

Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html http://ensiwiki.ensimag.fr/index.php/Asterisk's_external_configuration_(LDAP) Server:192.168.14.90 ?=> asterisk server:192.168.11.17 => ?ActiveDirectory Finally, this is my configuration file : [root at PB...

...'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on "Read the book online!" for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for thos...

...d every few days, I find myself sitting at the DID menu clicking the link that reads "Click here to get a random toll free number". I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/

...site for a while, I guess hoping I wouldn't be able to contact them. They never once responded to any of the telephone calls I made to them. I just signed in and see that they didn't ever refund my payment and close my account, either. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/

Hey there, I'm an asterisk newbie and have just joined this mailing list. I have to use asterisk as a call agent that supports MGCP requests. I'm reading the documentation from asteriskdocs and voip-info.org but those cover more specifically only IAX and SIP configuration. I'd really appreciate it if someone can tell me where to find more detailed documentation on how to configure asterisk to work with MGCP. Thanks