[asterisk-users] zap sending fax congested

hello all, i try to send a fax over a zap channel but it is not working i always get congested but receiving fax over the channel is working. here are my configs: zaptel.conf: # hfc-s pci a span definition loadzone=nl defaultzone=nl span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,2,3,ccs,ami bcha

[asterisk-users] dtmf and ivr

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,SetMusicOn

Re: [asterisk-users] 2 devices using same sip account

how isit possible to get the VM there when one line is busy? regards rene On Tue, 19 Dec 2006 09:48:01 -0800 Carla Schroder <[EMAIL PROTECTED]> wrote: Your phones only register once, when they first start up. Seems to me that having multiple phones on the same account is asking for trouble- wh

Re: [asterisk-users] 2 devices using same sip account

the lines get answered. It's easy enough to test. If it doesn't go to voicemail, then perhaps this is what you want: http://www.voip-info.org/wiki/view/Asterisk+tips+findme On Tuesday 19 December 2006 9:58 am, René Enskat wrote: how isit possible to get the VM there when one line i

Re: [asterisk-users] 2 devices using same sip account

ndme On Tuesday 19 December 2006 9:58 am, René Enskat wrote: > how isit possible to get the VM there when one line is busy? > > regards rene > > On Tue, 19 Dec 2006 09:48:01 -0800 > > Carla Schroder <[EMAIL PROTECTED]> wrote: > > Your phones only register once, wh

[asterisk-users] When line in use busy signal?!

hello all, how isit possible to give a busy signal if the line is in use? For me it is ringing and signaling on my phone, when i call out and i get another call. My hint is this: exten => 31,hint,SIP/1000131&SIP/1000131a i have one softphone an one hardphone Regards René -- René

[asterisk-users] cannot call out

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 s

[asterisk-users] pickup internal and external calls

hello, i want to make a dialplan where i can pickup calls to an extension when there are internal and external calls. i want to use only one prefix for pickup both situations so there is a plan how to check if the incoming call is an internal call or an extern??? regards rene _

[asterisk-users] Rxfax and txfax

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]:

[asterisk-users] pickup internal and external calls

hello, i want to make a dialplan where i can pickup calls to an extension when there are internal and external calls. i want to use only one prefix for pickup both situations so there is a plan how to check if the incoming call is an internal call or an extern??? regards rene _

[asterisk-users] Pickup failover

hello, i have configured internal pickup my problem is when an external call is coming i cannot pickup coz the extension is an external number. isit possible to pickup the external via n+101 prio or is ther any other solution? my config: exten => _*8.,1,GoToIf($["${CDR(userfield)}" = "EXTERN_INC

[asterisk-users] Hinting and Realtime

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

AW: [asterisk-users] Hinting and Realtime

2007 15:08 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Hinting and Realtime René Enskat wrote: > 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 unavailab

[asterisk-users] cutting hash in dial app

hello, isit possible to cut off the hash behind a dial string? coz we have a provider who gives us an error 600 "Declined" if ther is a hash in dial command. for example: Dial("SIP/x.x.x.x-b7d2d870", "SIP/[EMAIL PROTECTED] x") and i have to cut out: -b7d2d870 regar

[asterisk-users] cutting hash in dial app

<>___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [asterisk-users] 7970 sip success

Mmm i have set it in my MySQL Database in the row: Variables buggymwi = yes But can't see MWI Regards rene -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Zachary Whitley Gesendet: Freitag, 27. April 2007 00:09 An: Asterisk Use

[asterisk-users] asterisk-cluster with one database

hello all, somebody know how it is possible to have 2 asterisk and one database which is shared for both * ? i saw a problem with hinting that only 1 asterisk where the call ist made knowing that th eline is in use, the other said line is free. Is there a possibilty to let both * see that the line

[Asterisk-Users] App_rxfax problem

When i load the fax modules into the asterisk i got this errors but compile was ok! I have the latest cvs head    [res_musiconhold.so] => (Music On Hold Resource)  == Registered application 'MusicOnHold'  == Registered application 'WaitMusicOnHold'  == Registered application 'SetMusicOnHold'  ==

[Asterisk-Users] WG: App_rxfax problem

nobody has problems like me?   Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 08:35An: 'asterisk-users@lists.digium.com'Betreff: App_rxfax problem When i load the fax modules into the asterisk i got this errors but compile was ok!

AW: [Asterisk-Users] WG: App_rxfax problem

: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Giovanni Miano Gesendet: Donnerstag, 1. Dezember 2005 14:49 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem check /var/log/asterisk/full 2005/12/1, René Enskat [Teamware GmbH

AW: [Asterisk-Users] WG: App_rxfax problem

I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! --

[Asterisk-Users] WG: App_rxfax problem

  so is there a solution in the next cvs udpate? Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005 14:47An: 'asterisk-users@lists.digium.com'Betreff: WG: App_rxfax problem I just sent the error in full log: Dec 1 15:01:08 VERBOSE[27950

AW: [Asterisk-Users] WG: App_rxfax problem

But i have this in astewrisk log: Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so] Dec 1 15:01:08 WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: fax_set_phase_d_handler Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so failed! --

Re: [Asterisk-Users] WG: App_rxfax problem

Dunno :) what do you thing is wrong there? the compile was fine! I only need a solution how to fix this error!! On Sat, 03 Dec 2005 01:52:03 +0800 Steve Underwood <[EMAIL PROTECTED]> wrote: How could a CVS update fix an error you have made during installation? Steve René Enskat [Te

[Asterisk-Users] CDR Accounting Problem

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. So

[Asterisk-Users] SVN Revision 7230

hello,   I always update trough CVS from the cvs tree but i only see this revision 7230 in the asterisk all the days but the changelog say there are already newer versions. Did i updated wrong or is the revison wrong? ___ --Bandwidth and Colocation pr

[Asterisk-Users] app_md5.so compile problem

after cvs update i recompiled asterisk now i get this on loading: Dec 8 20:15:54 VERBOSE[25425] logger.c: [app_md5.so]Dec 8 20:15:54 WARNING[25425] loader.c: /usr/lib/asterisk/modules/app_md5.so: undefined symbol: option_priority_jumping Dec 8 20:15:54 WARNING[25425] loader.c: Loading mod

[Asterisk-Users] Hangup after dialing

i updated to actual sVN but now when i call with my phone i get a hangup when the clal should be ringing. with the branch all is fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update opt

[Asterisk-Users] CIDNUM CIDNAME

Does the CIDNUM and CIDNAME is not any longer working? How do i get the parts from the CALLERID? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[Asterisk-Users] ChefSec function

Somebody implemented the Chef-Secretary function in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Setting Language

Hey guys   Somebody can say how to set the language in the actual SVN release i tried alle pssible terms but nothing is working it tried:   exten => 3,1,Set(LANGUAGE()=de) exten => 3,1,SetLanguage(LANGUAGE()=de) exten => 3,1,Set(LANGUAGE=de)       -- Executing Set("SCCP/1000131-0006", "Langu

[Asterisk-Users] Goto after Dial PRoblem

i want to forward a call after the dial is not succesfull. But the problem is when the phone is not registered i get this error:   Dec 20 15:01:45 VERBOSE[15092] logger.c: -- Executing Set("SCCP/1000131-000b", "LANGUAGE()=de")Dec 20 15:01:45 VERBOSE[15092] logger.c: -- Executing Set(

[Asterisk-Users] WG: Goto after Dial PRoblem

somebody has a hint for my problem plz? This worked before but now it doesn't.   Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 20. Dezember 2005 15:05An: 'Asterisk Users Mailing List - Non-Commercial Discussion'Betreff: Goto after Dial PRo

[Asterisk-Users] Zaptel SVN

Hi, i can't compile the latest svn update from zaptel:   /lib/modules/2.6.14-1.1653_FC4smp/buildmake -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modulesmake[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686'  CC [M]  /usr/src/zaptel/zaptel.o/usr/src/zapte

[Asterisk-Users] Asterisk RELAY

Hello all,   I havbe a little problem here. I want to connect a SwyxPBX to the Asterisk. If i configure the swyx as a client all is fine but i want that the swyx server can call over the pbx without user authentication, the asterisk should see the IP and say ok this server can make a call over m

[Asterisk-Users] SVN Compile Error

build_tools/make_version_h > include/asterisk/version.h.tmpif 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.tmpif cmp -s .cleancount .lastclean ; th

[Asterisk-Users] pattern matching

Hi all.   I tried to build a pattenrmatching for a numberrange but the asterisk won't hear on it:   _49892351207[6-7][0-9]   if i make a:   _4989235120760   all is fine   Somebody has a hint fo rme? ___ --Bandwidth and Colocation provided by Easynews.

[Asterisk-Users] CDR Accounting Question

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. S

[Asterisk-Users] * Accounting with Oracle

Hello all, I use the asterisk with a oracle db in th ebackend. I want to use the db for accounting also. I saw that AMP has a mysql table with the accounting datas. Isit possible to por this to oracle or does anybody has a accounting agi or whatever which uses oracle? Regards Rene ___

[Asterisk-Users] Asterisk and NAT

Hey guys. I have to put my * behind a Firewall through nat on the firewall. The asterisk is running, but for example a register to an outside PSTN provider won't work. I enabled nat for the register but i only get Code 120 Send request. The other problem is, when i try to register with a sip phon

AW: [Asterisk-Users] Asterisk and NAT

Hi, Yes i opened 5060 and range -20001 The firewall is not blocking. I tried to set the externip and localnet but can't register to the pstn gateway and can't onnect with my nat phones. > -Ursprüngliche Nachricht- > Von: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Im Auftrag > vo

[Asterisk-Users] AGI Variable problem

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 get_variab

[Asterisk-Users] SetCallerID Problem

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", "4989427") in new stack -- Executing Se

Re: [Asterisk-Users] AGI Variable problem

> wrote: for some reason your script is not executing the get_var correctly, as you can see in the output, asterisk is saying: "invalid or unknown command". check the internals of your script, the most common reason is that you are mispelling the command. best regards On 10/13

[asterisk-users] WG: CDR ist getting wrong status

Hi, It seems the cdr modul always put ANSWERED Status into accounting table, even if it is not answered: Jul 11 12:29:47 DEBUG[18722] app_dial.c: Exiting with DIALSTATUS=CANCEL. Jul 11 12:29:47 VERBOSE[18722] logger.c: == Spawn extension (macro-call-cisco, s, 5) exited non-zero on 'SIP/1000131

[asterisk-users] realtime oracle dialplan select

somebody know a good way howto select datas from * oracle database inside the extensions? for mysql there are functions. are there for oracle similar ways?   regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users maili

[asterisk-users] mobile refusing call

Hi list,   I have a problem. I have an asterisk <--> Cisco Pots gateway. The problem is when i call via sip over the asterisk over the pots GW to a mobile phone and refuse th ecall on this mobile the sip phone is still ringing. it seems the cisco gw se on th eone site that the call ist busy/refu

[asterisk-users] WG: mobile refusing call

Hi,   Nobody has a hint for this? this seems to be a big problem when calling!   regards rene Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 6. September 2006 11:39An: 'Asterisk Users Mailing List - Non-Commercial Discussion'Betreff: mobile refusin

[Asterisk-Users] AGI Problem

Hmm still have problems with the get variable with PHP i have this error now separated with a script: Sending string GET VARIABLE CALLERIDNUM\n to Asterisk... Wroten bytes to STDOUT: 25 Reading 80 bytes response from Asterisk... Received response: 510 Invalid or unknown command __

AW: [Asterisk-Users] AGI Problem

ling List - Non-Commercial Discussion > Betreff: Re: [Asterisk-Users] AGI Problem > > Quoting "René Enskat [Teamware GmbH]" <[EMAIL PROTECTED]>: > > In my experience most AGI problems I had came from other info > sent to the terminal via verbose commands and othe

AW: AW: [Asterisk-Users] AGI Problem

, 17. Oktober 2005 12:29 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: AW: [Asterisk-Users] AGI Problem > > Quoting "René Enskat [Teamware GmbH]" <[EMAIL PROTECTED]>: > > What I normally do now with agi->verbose is to pass it a > vari

[Asterisk-Users] Asterisk hangs

Since some CVS Updates the asterisk hangs after command: reload or restart now. Then i have to kill -9 th eprocess. Nothing will be outout inside the CLI but i can type commands. Somebody know this problem? And the CallerID bug still seems to be in there too. Regards rene _

[Asterisk-Users] cvs core dump

Hi i checkout today the latest cvs but after that i get coredumps. Somebody know? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/list

[Asterisk-Users] NAT Problem after first call

Hey all,   I have little problem with my NAT clients on asterisk. After I called the clients one time where all is fine I try to call again and then the asterisk only say CALLED I have to reset the phone and reregister the phone so I can call again the phone. Somebody can help me?  

[Asterisk-Users] sKinny in database

Hi,   Isit possible to make the skinny working over a odbc/mysql/oracle db?   what i have to put in the extconfig.conf and how must the tables look like?   Hope somebody can help me..   thx rene ___ --Bandwidth and Colocation sponsored by Easynews.com

[Asterisk-Users] Skinny.conf and sccp.conf

  Hi,   I want to try the skinny/sccp protocol. Somebody can give me a working config for a cisco 7960 or 7970 ip phone? Isit possible to forward a SIP extension to the skinny phones? Coz i use normally a sip phone and i only want to forward this calls to the skinny phone.   regards Rene __

[Asterisk-Users] SCCP: ServiceURL and Mailbox Notification

  Hi. I tried to configure the ServiceURL on the asterisk inside the xml but i can't get it ro work i always get the errror hos tnot found and the ServiceURL field in the telephone is empty. I tried to put it in den SEPxx AND XmlDedault config without success. This is the url: http://phone-xml.b

[Asterisk-Users] COREDUMP in actual CVS

Actual cvs is impossible to start get coredump:   == Registered application 'SetRDNIS' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk)  == Parsing '/etc/asterisk/alarmreceiver.conf': Found  == Registered application 'AlarmReceiver' [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder

Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification

CCMCIP/authenticate.aspL> http://192.168.2.10/CCMCIP/xmldirectory.asp http://192.168.2.10/CCMCIP/GetTelecasterHelpText.aspnURL> http://192.168.2.20/CiscoServices/fetchPhoneObject 96 96 0 1 3804 192.168.2.10 # On Fri, 2005-11-04 at 15:14 +0100,

Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification

-04 at 15:14 +0100, René Enskat [Teamware GmbH] wrote: Hi. I tried to configure the ServiceURL on the asterisk inside the xml but i can't get it ro work i always get the errror hos tnot found and the ServiceURL field in the telephone is empty. I tried to put it in den SEPxx AND XmlDedau

[Asterisk-Users] Calling lines

hi guys,   Isit possbile to show busy lines from tthe asterisk to be shown on cisco phones at the function buttons? I have cisco 7970 (snom phones have the same) and i want to have some numbers at the keys and if this number ist talking i want to see that. Normal like isdn pbx in normal way with

[Asterisk-Users] hint problem

i enabled hint for some of my SIP and SCCP lines but on all i have Temp Fail and unavailable line status. when i make: SIP SHOW SUBSCRIPTIONS nothing is shown   call-limit = 2useclientcode=yesnotifyringing=yes   is in the config.   Somebody can help me?

[Asterisk-Users] Siemens OptiPoint 4xx

Somenbody know if the HINT function is not working for the OptiPoint4xx series?. I configured it but the keys are not working. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com htt

AW: [Asterisk-Users] Siemens OptiPoint 4xx

Hi stephen,   I have the latest SIP firmware form the siemens site! I have an OptiPoint 400 Standard. I configured the button with a targetnumber but if this line is talking no lights are on. And yes the sip firmware is on the unit.        Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auft

[Asterisk-Users] CDR Accounting PRoblem

I have a new 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 ansering our call and then forward to the pstn number. So i have billsecs all the time even it is only ringing or so.

[Asterisk-Users] Astfax with current CVS

Hi somebody can say me how i can integrate astfax1.0 in the current cvs tree? I installed astfax spandsp succesfully. But now i want to built the asterisk with the txfax and rxfax flag but the Makefile patch won't work anymore.. Somebody can help me?   Regards rene _

[Asterisk-Users] Astfax problem

ok got the patchfile to work but now i have compiling errors:   gcc  -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer  -fPIC   -c -o app_rxfax.o

[Asterisk-Users] Asterisk trunk cisco 2851

  Hi All,   Somebody here has experiences with asterisk server which trunks to a cisco 2851 via sip/h323. The cisco is the gatekeeper to the pstn network. Somebody has a sample configuration here for the cisco?   Regards rene ___ --Bandwidth and Coloc

[Asterisk-Users] calling between contexts

hi all,   somebody know a way how to call between contexts which are in a realtime database?   i tried to include them wise versa in extension.conf but this is not working. Is there another way?   regards rene ___ --Bandwidth and Colocation provided b

[asterisk-users] WG: CDR Accounting wrong

 Hi  * ,   I have the problem that the cdr account sthe ringing seconds too. normally it should begin accounting when the asterisk gets a answer but it seems it is accounting all the time since the sip connection from client to client is established.   I use Oracle DB with the cdr plugin from as

[Asterisk-Users] asterisk and DDI

Hi,   Somebody has some infos for asterisk and swyx connected via DDI? Somebody has a example config for ddi wiith asterisk?   regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or updat

AW: [Asterisk-Users] simple perl-agi - where's the error?

Tried: $DIALSTRING??? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Lenz Gesendet: Montag, 20. März 2006 12:56 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] simple perl-agi - where's the error?

[Asterisk-Users] Cisco 7970 SIP Image - hint lines

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? ___ --Bandwidth an

[Asterisk-Users] Realtime mapping problem after svn upgrade

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 availableMar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration fr

[Asterisk-Users] Asterisk dialing over asterisk to PSTN

hello all   soembody can give me an example config how can i let dial a asterisk server via SIP over another asterisk server to a pstn gateway ip?!?! asterisk1: x.x.x.x have to dial over asterisk2: y.y.y.y and then the asterisk2 should forward the call to the PSTN gateway. What i have to set in

[Asterisk-Users] Asterisk svn starting problem

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.sonoload => format_mp3.sonoload => format_pcm_alaw.so.sonoload => format_pcm_alaw.so   compiling was fine just some warnings   somebody has any i

[Asterisk-Users] Asterisk-addons compiling problem

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql   `ls *.c`make -C format_mp3 allmake[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

AW: [Asterisk-Users] Asterisk svn starting problem

ECTED] Im Auftrag von Dave Cotton Gesendet: Mittwoch, 5. April 2006 09:05 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] Asterisk svn starting problem On Wed, 2006-04-05 at 08:52 +0200, René Enskat [Teamware GmbH] wrote: > hi > > i updated aste

[Asterisk-Users] Realtime oracle compiling problem

I can'T compile my oracle realtime library any more i updatet the svn today and now i tried to recompile my oracle realtime driver and now it gives me that errors:     cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/oracle/10.1.0.4/client   -c -o res_config_oracle.o res_config_oracle.cres_co

[Asterisk-Users] 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 10:21:18 DEBUG[5873] loader.c: Unexpected signature: 8e 93 22 83 f5 c3 c0 75 ff 8b a9 be 7c 43 74 63Apr 10 10:21:18 WARNING[5873] loader.c: Unexpected key

[Asterisk-Users] WG: G729a error

  Somebody can say me what i can do that the g729 is working?   Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] Gesendet: Montag, 10. April 2006 10:21An: 'asterisk-users@lists.digium.com'Betreff: G729a error when i load asterisk i got this error and cant start * wit

[Asterisk-Users] sending special infoa fter login

hello all   Isit possible to send special informations to a phone after it registered? i want to send some config infos to the phone after it registered to the *.   Is that possible? And if yes how?   regards rene ___ --Bandwidth and Colocation provid

[Asterisk-Users] Quad ISDN card

Hi all,   Somebody know if the AVM C4 Quad ISDN card is supported by the current asterisk version?   regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http:/

[Asterisk-Users] SciTel Brix-QE card

Is this card compatible with asterisk? SciTel Brix-QE   Rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] pattern matching

hi all,   i want to build a extension that when i call 46-50 that ONE a account is ringing i have this:   exten => [46-50],1,Set(LANGUAGE()=de)exten => [46-50],2,CDR(userfield)=INTERNexten => [46-50],3,MusicOnHold(0.5)exten => [46-50],4,SIP/1000144|60|wWexten => [46-50],5,Hangup but it is not wo

AW: [Asterisk-Users] pattern matching

exten => [46-50],3,Answer > exten => [46-50],4,MusicOnHold(0.5) exten => > [46-50],5,SIP/1000144|60|wW exten => [46-50],6,Hangup > > René Enskat [Teamware GmbH] wrote: >> hi all, >> >> i want to build a extension that when i call 46-50 that ONE a account >

[Asterisk-Users] Realtime extension

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 se

[asterisk-users] 1.4 Beta and oracle

Morning all,   I updated to 1.4 now but it seems the oracle is not working with it? I get error with 1.2 all is fine:  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: Regist