search for: enskat

hi i updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload => format_au.so noload => format_mp3.so noload => format_pcm_alaw.so.so noload => format_pcm_alaw.so compiling was fine just some warnings somebody has any idea? -------------- next part -------------- An HTML attachment was scrubbed...

hello all, My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 regards rene -------------- next part -------------- An HTML attachment was scrubbed... URL:

We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",

My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack

I have a problem with the cdr. We terminate through a pstn provider to the pstn network. The problem is now the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn number. So i have billsecs all the time even it is only ringing or so. Somebody has a solution for that? -- Executing Dial("SIP/1000114-fcf8",

Hello I patche dmy 7970 with the current SIP image i have 2 lines on it via sip and 6 hint speeddials but it seems thats only a speeddial no infos about busy status or so comes to the speddial button. somebody can help me?

Somebody can say me what i can do that the g729 is working? _____ Von: Ren? Enskat [Teamware GmbH] [mailto:ren@teamware-gmbh.de] Gesendet: Montag, 10. April 2006 10:21 An: 'asterisk-users@lists.digium.com' Betreff: G729a error when i load asterisk i got this error and cant start * with the g729 codec: Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10...

...config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available Mar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available regards rene -- Ren? Enskat Internet-Administrator Teamware GmbH Stahlgruberring 11 D-81829 M?nchen Tel: 089-427005.31 Fax: 089-427005.55 E-Mail: ren@tmwr.de <http://www.tmwr.de/> http://www.tmwr.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com...

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi, i can't compile the latest svn update from zaptel: /lib/modules/2.6.14-1.1653_FC4smp/build make -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined

...Trixbox? > (Steve Sobol) > 10. Re: Multi Operator (Noc Phibee) > 11. Re: Linux distro + Asterisk or Trixbox? (Carla Schroder) > 12. Re: is it possible to use Asterisk voicemail as anouncement > system only? (Wilson Pickett) > 13. zap sending fax congested (Ren? Enskat) > 14. Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed > asterisk 1.2.13 (Jean-Yves Avenard) > 15. Re: Linux distro + Asterisk or Trixbox? (Vicky) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun...

build_tools/make_version_h > include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi

hi all. i upgraded my asterisk today via svn but now my oracle realtime is not longer working it always say: Mar 29 08:10:54 WARNING[3876] config.c: Realtime mapping for 'sippeers' found to engine 'oracle', but the engine is not available Mar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration from 'sip:xxx@xx.xxx-xxx.de' failed for xx.xx.xx.x- Username/auth name mismatch

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o common.o common.c gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations

i have realtime running over oracle database when i have some _ extensions in the database the asterisk won't accept them. Here i tried to call number 47. the extension for this one in the db is: _4[6-9] so the second select should found something with sqlnavigator i find the row but asterisk seems to stop continuing after that i get th emessage invalid extension. May 10 11:58:48

hello all, isit possible to do sth. so that if the sip-phone is in use and a call is incoming that the caller gets a busy signal? coz i wan to have the incoming party get a busy signal if i'm at phone. regards rene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061120/6f438f28/attachment.htm

hello, i try to build a IVR for our company my problem is that the dtmf tones are not recognized by the phones i tried several phones. BUT when i call the voicemail i can navigate with all phones through the menu. I use * 1.2 here is the context: [ivr] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ;SAI menu -

hello all. i switched to * 1.4 and have now 2 problems. 1. i can't make a call out with the current branch i always have in the logfile: [Jan 9 14:45:09] NOTICE[15246] chan_sip.c: Unable to create/find SIP channel for this INVITE With the asterisk 1.4 Release it is working, 2. when i do "core show hints" i see the channels on idle and unavailable BUT when a channel is active

somebody know how to compile the rxfax and txfax apps under asterisk 1.4.0?? i get this errors: Generating embedded module rules ... make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. make[1]: Nothing to be done for `all'. [CC] app_rxfax.c -> app_rxfax.o