similar to: asteriskdocs.org says 3rd ed. is latest

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 ones. Asterisk? ->

I'm having annoying errors trying to get configure working. tar xvzf /usr/local/src/asterisk-1.8.6.0.tar.gz cd asterisk-1.8.6.0 ./configure I get complaints related to pwlib / ptlib... checking for openr2_chan_new in -lopenr2... no checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking

Dear All, I want to play music in my AGI while i am searching for a field in DB. Actually during some processes in AGI i need to play music . Thanks in advanced. Regards, Mehdi

I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two guides. The first is the one from "Asterisk: The Definitive Guide" 4th edition (http://ofps.oreilly.com/titles/9781449332426/asterisk-Install.html) and the one from Billy Chia "How to Install Asterisk 11 on Ubuntu 12.04 LTS"

Hello I'd like to display CID information on users' monitor running Windows. I know I can run a script through the dialplan to send a datagram that is picked up Impulse Technology's free NetCID (www.imptec.com), but I'd rather use an open-source solution. An alternative would be to use a Windows application that would connect to Asterisk's AMI. I don't know if multiple

Is this command the best way to access a MySQL database - MYSQL(Connect connid dhhost dbuser dbpass dbname) ? I thought I heard that using ODBC was a bit more stable. Anyone have any experience? Thanks, Matt

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 @

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 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? *

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There 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

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

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 for 60 seconds more work it would be a bigger benefit to the asterisk

Having some issues with a TE131 apparently causing kernel panics. This is on CentOS 5.10 with DAHDI 2.9.1 (also tried 2.9.2-rc1, no difference.) There?s also an Tracebacks are very similar to this one: https://pbs.twimg.com/media/BrdrQf3CAAE3RXQ.jpg It?s configured with FXSKS signaling and the other end is an Adtran 850, though I?ve also experienced it crossed-over to an older TE13x doing

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:

Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and 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

Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: >My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable

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

In my attempt to setup a single FXS line I have been following the instructions for "Telephony Card Drivers" on the asteriskdocs.org site. I have managed to checkout, make and install the zaptel code and can load the zaptel module but when I attempt to load the wcfxs module it tells me: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install

Regarding my earlier post about labels and the 'n' priority: The TFOT book covers the use of these. See the box on page 81 entitled "Unnumbered Priorities." http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip -MC