similar to: call forward all except the extension it is forwarded to

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! 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

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

Hello, I'm just wondering if anyone has fixed the 'InUse&Ringing' problem. * v1.4.23.1 Ta

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open

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 it somehow possible to evaluate the SIP response code inside the dialplan? I have an Asterisk server which forwards requests to various PSTN gateways with SIP. If the Dial() attempt is not successful I want to differ at least these 3 options: - called destination is busy (486): e.g. activate auto-redial - called destination does not exist, unassigned number (404) - gateway is broken,

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hi, I have a question. How can I configure my sip.conf to make a SIP phone busy on incoming and outcoming calls? I explain my problem. When SIP phone receive a call and then I try to call that phone, I find it busy. When SIP phone make a call and I try to call that phone, I find it avaible and it rings but I want to find it busy. I configure sip.conf like following: [10] type=friend qualify=yes

I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, "the other Mark" in a truly Digium moment in Rostock, Germany on their way to listen to the sea shanties. http://tr.im/rawhide - be afraid, be very afraid (Adhearsions' Jason Goecke is also in the picture somewhere) /r

Hi All, I am using asterisk 1.4.19, I have setup the dialplans to get the incoming call and that will be sent to another context by local channel, In another context i have setup the ring group, that portion is working fine. I have noticed that when i have set one of the extension in call forwarding in phone (linksys) then it says to me 302 Moved Temporarily and call is forwarded to that number.

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!

Since 1.4.22 realtime status for sip peers seems to be broken. If I do a "sip show peers" from the CLI I get this: 2001/2001 192.168.2.234 D 5060 UNKNOWN Cached RT It is arbitrary which peers will say OK and which will say UNKNOWN and it changes over time. This is a problem with an application like the Asternic Flash panel because it uses the peer

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