similar to: Realtime extension

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.c res_config_oracle.c:53: warning: data definition has no type or storage class res_config_oracle.c: In

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

Dear friends, I cannot load the pachkages labdsv...which i do not understand is linked to 'mgcv' package. Anyway neither the last is loadable in my new R 2.10 version. Th emessage on the screen is the following: This is mgcv 1.6-1. For overview type `help("mgcv-package")'. Error in runif(1) : .Random.seed is not an integer vector but of type 'list' Error :

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",

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...

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 10:21:18

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

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?

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: Registration from 'sip:xxx@xx.xxx-xxx.de' failed for xx.xx.xx.x- Username/auth name

Hi! This teamware mailclient (teamware.com) that we use at the office has problems adding files as attachments from our Samba 3.0.2a share. The attachment file browser sees the files but fails to add them as attachments (with an error message: "valid.stf - file not found" regardless of the file name in question). I tracked this thing down to scandinavian letters (if your mail

Hi List, The below error caused the Asterisk to crash, if anyone have idea on this please reply,(Asterisk version :1.8.9) [Sep 2 15:59:53] WARNING[24418] channel.c: Codec mismatch on channel SIP/18202-0002d011 setting write format to ilbc from ulaw native formats 0x4 (ulaw) [Sep 2 15:59:53] WARNING[24418] channel.c: Unable to find a codec translation path from 0x4 (ulaw) to

Hi, Attached is a patch to log key comments upon successful login. It just adds them to the already existing verbose() call. I find it useful e.g. on shared accounts where it's sometimes not enough to have the key fingerprint in the log file. Can this be applied? -- Fernando Vezzosi qw(MDAx MTAw MDEw MDEx MDAw _5 MTEw _6 _1 _5 _5 _4 _2 _2 _6 MTEx _1 _5 _5 _4 _5 _2 _6

Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy

Here is the situation at hand, awaiting input from the collective minds of the asterisk-users list: 1. I have Asterisk registered as a SIP user agent to another Asterisk (which stands for the real provider, to another location). register => user:secret@10.0.0.1 2. I have a peer section for the provider. [provider] type=peer host=10.0.0.1 fromuser=user 3. I have two phones (FXS) on the

Hi, im trying to setup ASTCC but I'm getting it difficult. I've correctly set up the mysql database astcc and added a brand, trunk, route and a card as follows: brands +------+----------+------+--------------+------+--------+------+------+ | name | language | inc | publishednum | did | markup | days | fee | +------+----------+------+--------------+------+--------+------+------+ | FWD

I have some assembly code that uses .equ to set a symbol to a computed constant. It works in GCC but not LLVM. I am not sure about the correct terminology, but here is an example: .syntax unified .set size, end - start add r0, r0, #size start: .space 0x10 end: GCC will compile this to `add r0, r0, #16`. Compiling with llvm-mc gives: set.s:4:13: error: invalid operand for

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: