similar to: Parsing custom SIP headers

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

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

2009/1/30 Philipp Kempgen <philipp.kempgen at amooma.de> > Olivier schrieb: > > Here http://www.voip-info.org/wiki/view/Asterisk+database , you can > read: > > "Also, since it's a normal Berkely db1 (version185) file its contents can > be > > viewed/dumped with the standard *db1_dump185* tool. Thus db1_dump185 -p > > /var/lib/asterisk/astdb will

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

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

Hi, The descriptions of Progress() and Proceeding() are really vague. Progress(): ---cut---------------- [Synopsis] Indicate progress [Description] Progress(): This application will request that in-band progress information be provided to the calling channel. ---cut---------------- Proceeding(): ---cut---------------- [Synopsis] Indicate proceeding [Description] Proceeding(): This

Hello Philipp and All, My scenario is a bit different than the one I had explained before. I'm sorry. Let's suppose I have someone calling one of my Asterisk clients. This asterisk client has CFB (Call Forward Busy) activated. The forward number is a Voice Mail System, however is not a Asterisk's Voice Mail. It is a third party Voice Mail System, that has a SIP Trunk with my

Hi, Can somebody please shed some light on how to use the QUEUE_VARIABLES() function? The built-in help says ---cut--- Return Queue information in variables [Description] Makes the following queue variables available. QUEUEMAX maxmimum number of calls allowed QUEUESTRATEGY the strategy of the queue QUEUECALLS number of calls currently in the queue QUEUEHOLDTIME current average hold time

Does anybody else have the feeling that custom messages (Verbose(1,...)) do not stand out enough on the CLI? We're sending messages like "Extension 123 is unknown" to the output and that should tell the user why a call to 123 fails but users fre- quently crank up the verbosity to 3 or 10 so our messages go unnoticed in many cases. My idea was to use terminal escape sequences to make

O boy. SIP infrastructure is so flexible that basically nobody gets it right. :-) You could easily have 20 different SIP network elements (/servers /services). Even more. And we get at least 5 new SIP-RFCs per day. They're all trying to fix things which the previous specifications didn't address. :-) Philipp Kempgen -- AMOOCON 2009, May 4-5, Rostock / Germany ->

Hi, Sorry for this "newb" question (but maybe a newb question once in a while is ok): What's the current state about Asterisk handling DTMF features via SIP INFO (dtmfmode=info) even when the media path has been reinvited (canreinvite=yes) to go directly from one phone to another? Somewhat related to this suspended issue: https://issues.asterisk.org/view.php?id=14126 How widely

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

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

2008/12/2 Philipp Kempgen <philipp.kempgen at amooma.de> > Philipp Kempgen schrieb: > > Olivier schrieb: > > > >> How can you use Dial application M(x) option from extensions.ael ? > >> (As a reminder, this M(x) executes macro x when Dial called party > answers). > >> > >> It seems to me that asterisk keeps looking for this macro in >

2008/12/18 Philipp Kempgen <philipp.kempgen at amooma.de> > Olivier schrieb: > > > I unsuccessfully tried to download AstManProxy, clicking over download > > button in http://github.com/davetroy/astmanproxy/tree/master . > > It failed with "XML error". > > Try again. It works. You're right : now it works ! I can't explain why it didn't

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

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

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?

On Debian the Asterisk Makefile does /usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .; which results in a /etc/rc2.d/S10asterisk being written. I think S10 is too early. bind9 : S15 mysql : S19 zaptel: S20 ntp : S23 What bothers me most is that mysql is not up when asterisk starts. That's a bad thing if there are #execs in your config files and if the scripts rely on