similar to: Latest AstManProxy [SOLVED]

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 >

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, I unsuccessfully tried to download AstManProxy, clicking over download button in http://github.com/davetroy/astmanproxy/tree/master . It failed with "XML error". How can you download AstManProxy ? Has the project moved to somewhere else ? Have its features been deprecated and replaced by something embedded in Asterisk code or elsewhere ? Regrads -------------- next part

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, 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

This was an old thread http://lists.digium.com/pipermail/asterisk-users/2007-November/200539.html so I'm starting a new one. Tzafrir Cohen wrote: > On Thu, Nov 15, 2007 at 06:47:04PM +0100, Philipp Kempgen wrote: >> 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

Hi, Just to be sure: Is there a dialplan function in Asterisk that parses custom "name-addr"-style SIP headers for me? If I wanted to do it right the syntax name-addr *(SEMI generic-param) is quite complex to parse in the dialplan using nothing but CUT(). It's so easy to make false assumtions about angle brackets (< >), whitespace (LWS), quotes (") around the

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

Your right with this part But as I also have some knowldge on other parts but ms , *nix etc I know it is nowadays possible for almost every email client to correctly display html email. And be honest does it not read more easy if you have a nice font and some markup available? I know mailman is an old package and should be more flexible in handling and distributing html email. For standards:

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, 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

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

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

SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the

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

---cut------------------------------------------------- -= Info about function 'PP_EACH_USER' =- [Syntax] PP_EACH_USER(<string>|<exclude_mac>) [Synopsis] Generate a string for each phoneprov user [Description] Pass in a string, with phoneprov variables you want substituted in the format of %%%%{VARNAME}, and you will get the string rendered for each user in phoneprov

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

man asterisk (1.6.0-beta9) says: > -n Disable ANSI colors even on terminals capable of displaying them. But on the CLI (asterisk -n -r) any "core show application" or "core show function" leaks color escape sequences. Gr??e, Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied ->