similar to: Update Caller-ID after Dial()

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

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

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

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, 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

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

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

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

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

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

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 -

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 => {}

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! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!