similar to: the-asterisk-book.com online (unstable version)

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. 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?

Hi, Does anyone know how to include a file in AEL using the #include "filename" syntax in .conf files? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 -

Hi, If I have multiple kernel sources in /usr/src, e.g. linux-headers-2.6.26-1-686 linux-headers-2.6.26.custom.1 how does the Zaptel Makefile(?) know which one to pick? Is it a good approach to compile the kernel first and then compile Zaptel "manually" afterwards? Or should I rather put zaptel in /usr/src/modules and use fakeroot make-kpkg ... modules_image in the kernel

Hi, Is Asterisk supposed to evaluate #exec's in an #include'd file? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 -> http://www.amoocon.de --

Maybe it's just me, but I get the impression that Grandstream is quite uncooperative. We (and others) have asked them multiple times to make the call- pickup code ("**") configurable but either they don't understand the request or they're unwilling to do anything about it. http://forums.grandstream.com/node/2848 http://forums.grandstream.com/node/709 Unfortunately their

On starting Asterisk (1.4) I get a whole bunch of WARNING[5858]: pbx_ael.c:4040 ast_compile_ael2: Warning: file /etc/asterisk/extensions.ael, line 932-932: Empty Extension! I find it a bit disturbing that this message has a level of WARNING (instead of NOTICE maybe) because the extensions in question are empty on purpose. The only reason they exist are the hints. hint(SIP/3000) 3000 => {}

Can you confirm that currently there is no way to update the caller ID via the manager interface once the B leg is ringing or connected? Looks like this would be feasible with the functions introduced in https://issues.asterisk.org/view.php?id=8824 ("[patch] Remote (called) Party Identification - chan_sip & chan_skinny implementation"). Such functionality could be desirable in

Wow. My message made it to the list after more than 3 hours. Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Gesch?ftsf?hrer: Stefan Wintermeyer Handelsregister: Neuwied B 14998

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

1.6 UPGRADE.txt: > * If you use any interface for modifying voicemail aside from the built in > dialplan applications, then the option "pollmailboxes" *must* be set in > voicemail.conf for message waiting indication (MWI) to work properly. This > is because Voicemail notification is now event based instead of polling > based. The channel drivers are no longer

Hi, because many people contacted me about this the last couple of days and I guess most of them are on this list anyway: - Yes, our new German voiceprompts for Asterisk 1.4 are ready and can be downloaded at http://www.amooma.de/asterisk/service/deutsche- sprachprompts/ - Yes, we are in discussion with Digium about including them into the normal install process. I have no timeline but a

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hi All; Which is better (to have more stable or release versions) of zaptel, libpri and asterisk: to use cvs or svn? In case of using cvs, why I need to type: export CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot In other words: what is the use of pserver, anoncvs, ... with cvs checkout? Note: How can I know all the variables needed for cvs checkout so I might need to do

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi, Just wanted to let you know that we have just made our GPL toolkit "Gemeinschaft" available to the public. (Finally.) Mostly German for now - about half of the strings in the language strings file have been translated to English. I'm a software developer, not a marketing guy, so ... svn co https://svn.amooma.de/gemeinschaft/trunk gemeinschaft-trunk German readers: see

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

Hi all, Is there a way to keep track in Asterisk of which phones are online in realtime using some MySQL DB table for exemple, much like "sip show peers" does in the CLI? Regards, Ricardo.