[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:

[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 refusing call Hi list

[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

[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

[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

[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

[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 by

[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

[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 working.

AW: [Asterisk-Users] pattern matching

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 is ringing i have this: exten = [46-50],1,Set(LANGUAGE()=de) exten = [46-50],2,CDR

[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

[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] 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:

[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 provided

[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

[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 * with the g729 codec

[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 idea?

[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.o

AW: [Asterisk-Users] Asterisk svn starting problem

] 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 asterisk today via svn

[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

[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

[Asterisk-Users] asterisk and DDI

Hi, Somebody has someinfos forasterisk 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 update

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] CDR Accounting Question

I have aproblem 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.

[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.com

[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 ; then echo

[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

[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]

[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 PRoblem i want

[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

[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",

[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

[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:

[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

[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

[Asterisk-Users] CDR Accounting Problem

I have aproblem 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.

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 [Teamware GmbH

[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] logger.c

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!

[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! I have

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] 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 app_rxfax.cIn

[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' ==

AW: [Asterisk-Users] Siemens OptiPoint 4xx

Hi stephen, I have the latest SIPfirmware 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 Auftrag

[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

[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

[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] 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] 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:

[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

# On Fri, 2005-11-04 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

[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] 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] 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 clientid I have to reset the phone and reregister the phone so I can call again the phone. Somebody can help

[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

[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] 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

-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 other stdout output. There is some info on the voip-info wiki about

AW: AW: [Asterisk-Users] AGI Problem

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 variable using print_r($outputvariable, true). thus if I want to output

[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 ?php

[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

Re: [Asterisk-Users] AGI Variable problem

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 10/13/05, René Enskat [Teamware GmbH] [EMAIL

[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

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 von

[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