[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: Registration from 'sip:[EMAIL PROTECTED]' failed for xx.xx.xx.x- Username/auth name mismatch Mar 29 08:10:58 WARNING[3876] 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 Mar 29 08:10:58 WARNING[3876] 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 --René EnskatInternet-Administrator Teamware GmbHStahlgruberring 11D-81829 München Tel: 089-427005.31Fax: 089-427005.55E-Mail: [EMAIL PROTECTED]http://www.tmwr.de ___ --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] 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, 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/refused but on the gw->sip side the cal is still active! somebody has a solution or hint for me? Thx! regards 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] 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/refused but on the gw->sip side the cal is still active! somebody has a solution or hint for me? Thx! regards 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] 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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/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 'SIP/1000131-093bd318' in macro 'call-cisco' Jul 11 12:29:47 VERBOSE[18722] logger.c: == Spawn extension (macro-call-cisco, s, 5) exited non-zero on 'SIP/1000131-093bd318' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '"4989xxx" <31>' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '31' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '089...' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '10001' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/1000131-093bd318' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/x.x.x.x-093cf108' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'Dial' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'SIP/[EMAIL PROTECTED]|60' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:41' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:41' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '2006-07-11 12:29:47' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '6' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '6' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'ANSWERED' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'DOCUMENTATION' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '146' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is '1152613781.34' Jul 11 12:29:47 DEBUG[18722] pbx.c: Function result is 'EXTERN_OUTGOING' ___ --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] 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 and when i have only ringing no answer i saw in the cdr tables billed seconds. somebody can confirm? regards 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] 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 Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 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] 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 seems to stop continuing after that i get th emessage invalid extension. May 10 11:58:48 DEBUG[18202] res_config_oracle.c: Oracle RealTime: Retrieve SQL: SELECT * FROM ast_extension WHERE exten = '47' AND context = '10001' AND priority = '1'May 10 11:58:48 DEBUG[18202] res_config_oracle.c: Oracle RealTime: Reconnected successfully.May 10 11:58:48 DEBUG[18202] res_config_oracle.c: Oracle RealTime: Retrieve SQL: SELECT * FROM ast_extension WHERE exten LIKE '\_%' ESCAPE '\' AND context = '10001' AND priority = '1' ORDER BY extenMay 10 11:58:48 DEBUG[18202] res_config_oracle.c: Oracle RealTime: Reconnected successfully. ___ --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] pattern matching
It seems it must be in thix way: _4[6-9] But this is not very confortable if you have 4x and 5x numbers :) -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Alasdair Gow Gesendet: Mittwoch, 10. Mai 2006 11:54 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] pattern matching What do you see on the asterisk console? do you see it setting the language etc or does it not match the pattern? > try > > exten => [46-50],1,Set(LANGUAGE()=de) > exten => [46-50],2,CDR(userfield)=INTERN 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 >> is ringing i have this: >> >> exten => [46-50],1,Set(LANGUAGE()=de) exten => >> [46-50],2,CDR(userfield)=INTERN exten => [46-50],3,MusicOnHold(0.5) >> exten => [46-50],4,SIP/1000144|60|wW exten => [46-50],5,Hangup but it >> is not working. >> - >> --- >> >> ___ >> --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 >> > > -- Regards, Alasdair Gow BSc (Hons) Support Specialist Colloquium Internet Support ___ --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 ___ --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 working. ___ --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] 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: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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: 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 returned by module /usr/lib/asterisk/modules/codec_g729a.soApr 10 10:21:18 WARNING[5873] loader.c: 1 error loading module /usr/lib/asterisk/modules/codec_g729a.so, abortedApr 10 10:21:18 WARNING[5873] loader.c: Loading module codec_g729a.so failed! ___ --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] 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 returned by module /usr/lib/asterisk/modules/codec_g729a.soApr 10 10:21:18 WARNING[5873] loader.c: 1 error loading module /usr/lib/asterisk/modules/codec_g729a.so, abortedApr 10 10:21:18 WARNING[5873] loader.c: Loading module codec_g729a.so failed! ___ --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] 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_config_oracle.c:53: warning: data definition has no type or storage classres_config_oracle.c: In function 'realtime_oracle':res_config_oracle.c:109: warning: incompatible implicit declaration of built-in function 'snprintf'res_config_oracle.c:127: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:129: warning: pointer targets in passing argument 3 of 'OCIStmtPrepare' differ in signednessres_config_oracle.c:130: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:133: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:138: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:142: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:144: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:146: warning: pointer targets in passing argument 1 of 'strncpy' differ in signednessres_config_oracle.c:146: warning: pointer targets in passing argument 2 of 'strncpy' differ in signednessres_config_oracle.c:168: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:174: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:180: warning: pointer targets in assignment differ in signednessres_config_oracle.c:188: warning: pointer targets in passing argument 1 of 'ast_variable_new' differ in signednessres_config_oracle.c:193: warning: pointer targets in passing argument 1 of 'ast_variable_new' differ in signednessres_config_oracle.c: In function 'realtime_multi_oracle':res_config_oracle.c:272: warning: incompatible implicit declaration of built-in function 'snprintf'res_config_oracle.c:295: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:297: warning: pointer targets in passing argument 3 of 'OCIStmtPrepare' differ in signednessres_config_oracle.c:298: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:301: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:306: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:310: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:312: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:314: warning: pointer targets in passing argument 1 of 'strncpy' differ in signednessres_config_oracle.c:314: warning: pointer targets in passing argument 2 of 'strncpy' differ in signednessres_config_oracle.c:336: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:342: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:354: warning: pointer targets in assignment differ in signednessres_config_oracle.c:361: warning: pointer targets in passing argument 2 of 'strcmp' differ in signednessres_config_oracle.c:364: warning: pointer targets in passing argument 1 of 'ast_variable_new' differ in signednessres_config_oracle.c: In function 'update_oracle':res_config_oracle.c:408: warning: incompatible implicit declaration of built-in function 'snprintf'res_config_oracle.c:427: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:430: warning: pointer targets in passing argument 3 of 'OCIStmtPrepare' differ in signednessres_config_oracle.c:431: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:435: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:439: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c: In function 'config_oracle':res_config_oracle.c:478: warning: incompatible implicit declaration of built-in function 'snprintf'res_config_oracle.c:490: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:492: warning: pointer targets in passing argument 3 of 'OCIStmtPrepare' differ in signednessres_config_oracle.c:493: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:496: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessres_config_oracle.c:500: warning: pointer targets in passing argument 3 of 'checkerr' differ in signednessr
AW: [Asterisk-Users] Asterisk svn starting problem
It said that these modules are to old for the version it seems asterisk has that now builtin or? I deleted the modules then it is working So asterisk brings these module sbuiltin and i don'T need asterisk-addons? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 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? And make install didn't mention anything about /usr/lib/asterisk/modules? -- Dave Cotton <[EMAIL PROTECTED]> ___ --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 ___ --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] 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 common.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o dct64_i386.o dct64_i386.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o decode_ntom.o decode_ntom.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o layer3.o layer3.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o tabinit.o tabinit.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o interface.o interface.cgcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -c -o format_mp3.o format_mp3.cformat_mp3.c:46: Fehler: Redefinition von »struct ast_filestream«format_mp3.c: In Funktion »load_module«:format_mp3.c:336: Warnung: Übergabe des Arguments 1 von »ast_format_register« von inkompatiblem Zeigertypformat_mp3.c:336: Fehler: zu viele Argumente für Funktion »ast_format_register«make[1]: *** [format_mp3.o] Fehler 1make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3'make: *** [format_mp3/format_mp3.so] Fehler 2 can't compile 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] 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? ___ --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] 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 sip.conf that asterisk1 can dial over asterisk2? regards 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] 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 from 'sip:[EMAIL PROTECTED]' failed for xx.xx.xx.x- Username/auth name mismatchMar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not availableMar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not availableMar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not availableMar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available somebody can say me whats wrong? ___ --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] 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 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] 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?
Try setting it to sth like SIP/200 instead of a single number.
l.
On Mon, 20 Mar 2006 11:56:50 +0100, Christian B <[EMAIL PROTECTED]>
wrote:
> Hello!
>
> I'm trying to setup a perl-deadagi, but my perl skills lack. can
> someone tell me why the following code doesn't work:
>
> #!/usr/bin/perl
> use Asterisk::AGI;
> $AGI = new Asterisk::AGI;
>
> $dialstring = $AGI->get_variable("DIALSTRING"); $res =
> $AGI->exec("DIAL $dialstring");
>
>
> the asterisk output says:
>
> AGI Rx << GET VARIABLE DIALSTRING
> AGI Tx >> 200 result=1 (089324154332)
> AGI Rx << EXEC DIAL ""
> -- AGI Script Executing Application: (DIAL) Options: () Mar 20
> 11:46:02 WARNING[21970]: app_dial.c:773 dial_exec_full: Dial requires
> an argument (technology/number) AGI Tx >> 200 result=-1
> -- AGI Script agirouter/dialscript.pl completed, returning 0
>
>
> so the get_variable-command seems to work, also the exec(with
> "$dialstring = 089324154332" the call goes out), but not setting the
> variable. should be so simple :-( astcc-agi seems to use the same
> syntax, so i have no clue what is wrong in my place...
> any ideas? thx!
>
> kind regards
> christian
>
--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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[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 update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.
Somebody has a
solution for that?
-- Executing Dial("SIP/1000114-fcf8", "SIP/[EMAIL PROTECTED]|60")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/connect.xxx.de-c61d is making progress passing it to
SIP/1000114-fcf8 -- SIP/connect.xxx.de-c61d answered
SIP/1000114-fcf8 -- Attempting native bridge of
SIP/1000114-fcf8 and SIP/connect.xxx.de-c61d
Regards
rene
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[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 mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ; else \ make clean; cp -f .cleancount .lastclean;\fi build_tools/make_defaults_h > defaults.h.tmpif cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \fi rm -f defaults.h.tmpfor x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; donemake[1]: Entering directory `/usr/src/asterisk/res'../build_tools/mkdep -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 -DT38_SUPPORT -DZAPATA_MOH -DOPENSSL_NO_KRB5 -fPIC `ls *.c`make[1]: Leaving directory `/usr/src/asterisk/res'make[1]: Entering directory `/usr/src/asterisk/channels'../build_tools/mkdep -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 -DT38_SUPPORT -Wno-missing-prototypes -Wno-missing-declarations -DZAPATA_PRI -DIAX_TRUNKING -DCRYPTO -fPIC `ls *.c`make[1]: Leaving directory `/usr/src/asterisk/channels'make[1]: Entering directory `/usr/src/asterisk/pbx'../build_tools/mkdep -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 -DT38_SUPPORT -fPIC `ls *.c`make[1]: Leaving directory `/usr/src/asterisk/pbx'make[1]: Entering directory `/usr/src/asterisk/apps'Makefile:53: *** missing separator. Schluss.make[1]: Leaving directory `/usr/src/asterisk/apps'make: *** [depend] Fehler 1 NEWEST SVN Checkout ___ --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] 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 me. Somebody know how to configure this on the Asterisk? regards 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] 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/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined/usr/src/zaptel/zaptel.c:224: Warnung: »fcstab« definiert, aber nicht verwendet/usr/src/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined CC [M] /usr/src/zaptel/tor2.o CC [M] /usr/src/zaptel/torisa.o/usr/src/zaptel/torisa.c:1145: Warnung: »set_tor_base« definiert, aber nicht verwendet CC [M] /usr/src/zaptel/wcusb.o CC [M] /usr/src/zaptel/wcfxo.o CC [M] /usr/src/zaptel/wctdm.o CC [M] /usr/src/zaptel/wctdm24xxp.o CC [M] /usr/src/zaptel/ztdynamic.o CC [M] /usr/src/zaptel/ztd-eth.o/usr/src/zaptel/ztd-eth.c:185: Warnung: Initialisierung von inkompatiblem Zeigertyp CC [M] /usr/src/zaptel/wct1xxp.o CC [M] /usr/src/zaptel/wct4xxp.o/usr/src/zaptel/wct4xxp.c: In Funktion »t4_interrupt«:/usr/src/zaptel/wct4xxp.c:2219: nicht implementiert: »inline« beim Aufruf von »__t4_framer_interrupt« gescheitert: function body not available/usr/src/zaptel/wct4xxp.c:2251: nicht implementiert: von hier aufgerufenmake[2]: *** [/usr/src/zaptel/wct4xxp.o] Fehler 1make[1]: *** [_module_/usr/src/zaptel] Fehler 2make[1]: Leaving directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686'make: *** [linux26] Fehler 2 ___ --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] 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 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("SCCP/1000131-000b",
"CDRUserField=INTERN")Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Set("SCCP/1000131-000b",
"MusicOnHold")Dec 20 15:01:45 WARNING[15092] pbx.c: Ignoring entry
'MusicOnHold' with no = (and not last 'options' entry)Dec 20 15:01:45
VERBOSE[15092] logger.c: -- Executing
Dial("SCCP/1000131-000b", "SIP/10001233|30")Dec 20 15:01:45 DEBUG[15092]
db.c: Unable to find key '10001233' in family 'SIP/Registry'Dec 20 15:01:45
DEBUG[15092] db.c: Unable to find key '10001233' in family 'SIP/Registry'Dec
20 15:01:45 NOTICE[15092] app_dial.c: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)Dec 20 15:01:45 VERBOSE[15092]
logger.c: == Everyone is busy/congested at this time
(1:0/0/1)Dec 20 15:01:45 DEBUG[15092] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing
VoiceMail("SCCP/1000131-000b", "[EMAIL PROTECTED]")Dec 20 15:01:45 WARNING[15092]
app_voicemail.c: No entry in voicemail config file for '233'Dec 20 15:01:45
VERBOSE[15092] logger.c: -- Executing
Hangup("SCCP/1000131-000b", "")Dec 20 15:01:45 VERBOSE[15092]
logger.c: == Spawn extension (10001, 233, 106) exited non-zero on
'SCCP/1000131-000b'Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Set("SCCP/1000131-000b",
"LANGUAGE()=de") in new stackDec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Playback("SCCP/1000131-000b",
"goodbye") in new stackDec 20 15:01:45 VERBOSE[15092]
logger.c: -- Playing 'goodbye' (language 'de')Dec 20
15:01:46 VERBOSE[15092] logger.c: -- Executing
Hangup("SCCP/1000131-000b", "") in new stackDec 20 15:01:46
VERBOSE[15092] logger.c: == Spawn extension (10001, h, 3) exited
non-zero on 'SCCP/1000131-000b'
But my dialplan
shows this:
...
4
Dial SIP/10001233|30<from here it jumps to 107
mailbox but this is only for busy but this is a unavailable
situation
5
Goto 10001|23|1
6
Hangup
105 VoiceMail [EMAIL PROTECTED]
Somebody can help me
here why the GOTO is not followed?
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[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("SCCP/1000131-000b",
"CDRUserField=INTERN")Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Set("SCCP/1000131-000b",
"MusicOnHold")Dec 20 15:01:45 WARNING[15092] pbx.c: Ignoring entry
'MusicOnHold' with no = (and not last 'options' entry)Dec 20 15:01:45
VERBOSE[15092] logger.c: -- Executing
Dial("SCCP/1000131-000b", "SIP/10001233|30")Dec 20 15:01:45 DEBUG[15092]
db.c: Unable to find key '10001233' in family 'SIP/Registry'Dec 20 15:01:45
DEBUG[15092] db.c: Unable to find key '10001233' in family 'SIP/Registry'Dec
20 15:01:45 NOTICE[15092] app_dial.c: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)Dec 20 15:01:45 VERBOSE[15092]
logger.c: == Everyone is busy/congested at this time
(1:0/0/1)Dec 20 15:01:45 DEBUG[15092] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing
VoiceMail("SCCP/1000131-000b", "[EMAIL PROTECTED]")Dec 20 15:01:45 WARNING[15092]
app_voicemail.c: No entry in voicemail config file for '233'Dec 20 15:01:45
VERBOSE[15092] logger.c: -- Executing
Hangup("SCCP/1000131-000b", "")Dec 20 15:01:45 VERBOSE[15092]
logger.c: == Spawn extension (10001, 233, 106) exited non-zero on
'SCCP/1000131-000b'Dec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Set("SCCP/1000131-000b",
"LANGUAGE()=de") in new stackDec 20 15:01:45 VERBOSE[15092]
logger.c: -- Executing Playback("SCCP/1000131-000b",
"goodbye") in new stackDec 20 15:01:45 VERBOSE[15092]
logger.c: -- Playing 'goodbye' (language 'de')Dec 20
15:01:46 VERBOSE[15092] logger.c: -- Executing
Hangup("SCCP/1000131-000b", "") in new stackDec 20 15:01:46
VERBOSE[15092] logger.c: == Spawn extension (10001, h, 3) exited
non-zero on 'SCCP/1000131-000b'
But my dialplan
shows this:
...
4
Dial SIP/10001233|30
[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", "Language()=de") --
Executing Answer("SCCP/1000131-0006", "") -- Executing
NoCDR("SCCP/1000131-0006", "") -- Executing
Wait("SCCP/1000131-0006", "1") -- Executing
VoicemailMain("SCCP/1000131-0006", "[EMAIL PROTECTED]") -- Playing
'vm-login' (language 'en')
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[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] 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/mailman/listinfo/asterisk-users
[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 options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 module app_md5.so failed! ___ --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] 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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.
Somebody has a
solution for that?
-- Executing Dial("SIP/1000114-fcf8", "SIP/[EMAIL PROTECTED]|60")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/connect.xxx.de-c61d is making progress passing it to
SIP/1000114-fcf8 -- SIP/connect.xxx.de-c61d answered
SIP/1000114-fcf8 -- Attempting native bridge of
SIP/1000114-fcf8 and SIP/connect.xxx.de-c61d
Regards
rene
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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] wrote: so is there a solution in the next cvs udpate? *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 14:47 *An:* '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: [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! ---- *Von:* René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] *Gesendet:* Donnerstag, 1. Dezember 2005 08:35 *An:* '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 the latest cvs head [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe ___ --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 ___ --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 ___ --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] 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! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Andrew Furey Gesendet: Freitag, 2. Dezember 2005 14:36 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem > > Ouch ... error while writing audio data: : Broken pipe > > If you are talking about the Ouch message, yes lots of people have > seen the error and its usually the result of some misconfiguration in > one of your files (likely zapata.conf). Correct me if I'm wrong, but isn't that message from mpg123 itself? It appears in the binary (via strings), and I've seen it at non-asterisk times too. AFAIK it comes up whenever the parent application (asterisk in this case) quits without closing it properly (hence, "broken pipe"). As such, this means that the above error simply shows that asterisk crashed (which they presumably already knew), and has nothing to do with the problem itself... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ --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 ___ --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] 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: [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! 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 the latest cvs head [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --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] 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! -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Rich Adamson Gesendet: Donnerstag, 1. Dezember 2005 15:05 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] WG: App_rxfax problem > nobody has problems like me? > --- > == Registered application 'StartMusicOnHold' > == Registered application 'StopMusicOnHold' > [app_rxfax.so]Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Ouch ... error while writing audio data: : Broken pipe Ouch ... error > while writing audio data: : Broken pipe Ouch ... error while writing > audio data: : Broken pipe ---End of Original Message- If you are talking about the Ouch message, yes lots of people have seen the error and its usually the result of some misconfiguration in one of your files (likely zapata.conf). Since you didn't provide anything reasonable for anyone to look at or comment on, its impossible to guess at what you might have done. The message would suggest that musiconhold probably has something to do with the problem because of the "flexibel rate" message. ___ --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 ___ --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] WG: App_rxfax problem
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! -Ursprüngliche Nachricht- Von: [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] <[EMAIL PROTECTED]>: > > nobody has problems like me? > > > > > ________ > Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED] > Gesendet: Donnerstag, 1. Dezember 2005 08:35 > An: '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 the latest cvs head > > [res_musiconhold.so] => (Music On Hold Resource) > == Registered application 'MusicOnHold' > == Registered application 'WaitMusicOnHold' > == Registered application 'SetMusicOnHold' > == Registered application 'StartMusicOnHold' > == Registered application 'StopMusicOnHold' > [app_rxfax.so]Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Warning, flexibel rate not heavily tested! > Ouch ... error while writing audio data: : Broken pipe Ouch ... error > while writing audio data: : Broken pipe Ouch ... error while writing > audio data: : Broken pipe > ___ > --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 > > > -- Giovanni Miano ___ --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 ___ --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] 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 the latest cvs head [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --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] 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' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' [app_rxfax.so]Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Warning, flexibel rate not heavily tested!Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --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] 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 file included from app_rxfax.c:15:../include/asterisk/file.h:55: Fehler: syntax error before »*« token../include/asterisk/file.h:55: Warnung: Funktionsdeklaration ist kein Prototyp../include/asterisk/file.h:56: Fehler: syntax error before »*« token../include/asterisk/file.h:56: Warnung: Funktionsdeklaration ist kein Prototypapp_rxfax.c: In Funktion »phase_e_handler«:app_rxfax.c:77: Warnung: implizite Deklaration der Funktion »fax_get_transfer_statistics«app_rxfax.c:78: Warnung: implizite Deklaration der Funktion »fax_get_far_ident«app_rxfax.c:79: Warnung: implizite Deklaration der Funktion »fax_get_local_ident«app_rxfax.c: In Funktion »rxfax_exec«:app_rxfax.c:189: Warnung: Zeigerziele bei Übergabe des Arguments 1 von »__builtin_strncpy« unterscheiden sich im Vorzeichenbesitzapp_rxfax.c:259: Warnung: Übergabe des Arguments 1 von »fax_init« von inkompatiblem Zeigertypapp_rxfax.c:260: Fehler: »t30_state_t« hat kein Element namens »verbose«app_rxfax.c:263: Warnung: implizite Deklaration der Funktion »fax_set_local_ident«app_rxfax.c:266: Warnung: implizite Deklaration der Funktion »fax_set_header_info«app_rxfax.c:267: Warnung: implizite Deklaration der Funktion »fax_set_rx_file«app_rxfax.c:269: Warnung: implizite Deklaration der Funktion »fax_set_phase_d_handler«app_rxfax.c:270: Warnung: implizite Deklaration der Funktion »fax_set_phase_e_handler«app_rxfax.c:281: Warnung: implizite Deklaration der Funktion »fax_rx_process«app_rxfax.c:284: Warnung: implizite Deklaration der Funktion »fax_tx_process«app_rxfax.c:321: Warnung: Übergabe des Arguments 1 von »fax_release« von inkompatiblem Zeigertypmake[1]: *** [app_rxfax.o] Fehler 1make[1]: Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Fehler 1 ___ --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] 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 ___ --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] 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.
Somebody has a
solution for that?
-- Executing Dial("SIP/1000114-fcf8", "SIP/[EMAIL PROTECTED]|60")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/connect.xxx.de-c61d is making progress passing it to
SIP/1000114-fcf8 -- SIP/connect.xxx.de-c61d answered
SIP/1000114-fcf8 -- Attempting native bridge of
SIP/1000114-fcf8 and SIP/connect.xxx.de-c61d
Regards
rene
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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 Auftrag von Stephen ArulrajGesendet: Sonntag, 27. November 2005 00:33An: Asterisk Users Mailing List - Non-Commercial DiscussionBetreff: Re: [Asterisk-Users] Siemens OptiPoint 4xx RenéCan you email the exact configurations? By the way, do you have the SIP application software for these phones and what version is it? Previously I was looking for the app/firmware for the latest optiPoint 410 and 420 series. Anyway I now get them preloaded in these sexy looking phones and they are being shiped out with the latest firmware (Ver. 4.1.59) from our company. It works great with the Asterisk. If you are interested, you can buy them pre-loaded with the firmware. Just visit these links and click to inquire to order them. http://www.solomonstar.com/listings/171.htmlhttp://www.solomonstar.com/listings/170.htmlBest,StephenRené Enskat [Teamware GmbH] wrote: 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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] 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 http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 system-telephones. regards rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
Hmm i tried your config but the service url ist still not working.
i have the 7.1 images on the phone.
and the message waiting icon is nothing there too but i have a new message on
the server
On Fri, 04 Nov 2005 11:35:32 -0600
Greg Oliver <[EMAIL PROTECTED]> wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
Default
CMLocal
M/D/Y
Central Standard/Daylight Time
2002
2001
2000
2427
2428
192.168.2.10
2002
2001
2000
2427
2428
192.168.2.11
Disable
Disable
false
2000
2000
2000
false
-1
Default
Default
120
TERM70.7-0-2-0S
{21ECCF08-13DB-4EC5-8BCE-B177569C489B}
English_United_States
1
en
4.1(3)
iso-8859-1
United_States
United_States
64
4.1(3)
1
0
http://192.168.2.10/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, 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 XmlDedault config without success.
This is the url:
http://phone-xml.berbee.com/menu.xml
In my old 7960 i always get a lettersymbol at my line when i got a
mailboxmessage via SIP but this won'z be with the sccp protocol?
Or how cna i have this symbols there?
I have new voicemessages on my asterisk but the telephone is saying
nothing about that.
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Re: [Asterisk-Users] SCCP: ServiceURL and Mailbox Notification
I use the newest from cisco site i think 7.1 or so does it matter for the xml?
On Fri, 04 Nov 2005 11:37:41 -0600
Greg Oliver <[EMAIL PROTECTED]> wrote:
Forgot to mention - it is 7.0.2-0S firmware
On Fri, 2005-11-04 at 11:35 -0600, Greg Oliver wrote:
I had the same issue. Here is a full config from a 4.1.3SR1 CCM for a
7970 - let me knwo if you need any others and I will tftp them off.
Thanks,
Greg
#
[EMAIL PROTECTED] ~ $ cat SEP00127F027D17.cnf.xml
Default
CMLocal
M/D/Y
Central Standard/Daylight Time
2002
2001
2000
2427
2428
192.168.2.10
2002
2001
2000
2427
2428
192.168.2.11
Disable
Disable
false
2000
2000
2000
false
-1
Default
Default
120
TERM70.7-0-2-0S
{21ECCF08-13DB-4EC5-8BCE-B177569C489B}
English_United_States
1
en
4.1(3)
iso-8859-1
United_States
United_States
64
4.1(3)
1
0
http://192.168.2.10/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, 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 XmlDedault config without success.
>
> This is the url:
> http://phone-xml.berbee.com/menu.xml
>
>
> In my old 7960 i always get a lettersymbol at my line when i got a
> mailboxmessage via SIP but this won'z be with the sccp protocol?
> Or how cna i have this symbols there?
>
> I have new voicemessages on my asterisk but the telephone is saying
> nothing about that.
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[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) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [app_math.so] => (Basic Math Functions) == Registered application 'Math' [skipping chan_modem_i4l.so] [app_sendtext.so] => (Send Text Applications) == Registered application 'SendText' [app_muxmon.so]Ouch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipeOuch ... error while writing audio data: : Broken pipe ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.berbee.com/menu.xml In my old 7960 i always get a lettersymbol at my line when i got a mailboxmessage via SIP but this won'z be with the sccp protocol? Or how cna i have this symbols there? I have new voicemessages on my asterisk but the telephone is saying nothing about that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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? Regards rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] AGI Problem
Hmm sorry can't follow you in the way.
You can say me how i have to change my script to that what do you mea?
#!/usr/bin/php -q
get_variable("SIPUSER");
if ($ID["result"] == 0) {
$agi->verbose("SIPUSER not set -- nothing to do");
exit(1);
}
$number = $ID["data"];
$agi->set_variable("MSN", exec("/var/lib/asterisk/agi-bin/msn4sip 111
222 333 $number"));
?>
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag von Obelix
> Gesendet: Montag, 17. Oktober 2005 12:29
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: Re: AW: [Asterisk-Users] AGI Problem
>
> Quoting "René Enskat [Teamware GmbH]" <[EMAIL PROTECTED]>:
>
> What I normally do now with agi->verbose is to pass it a
> variable using print_r($outputvariable, true).
>
> thus if I want to output a string "" or an array of some
> sort it goes out in the form
>
> $output = ""
>
> $agi->verbose(print_r($output, true))
>
> What I suggest now is to suppress screen output as much as
> you can and see if the 510 errors go away.
>
> I also realised after using phpagi 1 before that the variable
> hashes had changed in phpagi 2. So if you are adapting some
> code from phpagi 1 check the hashes. Do a print_r on the
> result variables and see if the hashes are what you expect them to be.
>
> > I have the phpagi 2 library too.
> > So what did you change in details there to mute the vebrose things?
> >
> >
> >
> >
> > > -Ursprüngliche Nachricht-
> > > Von: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] Im Auftrag von
> > > Obelix
> > > Gesendet: Montag, 17. Oktober 2005 11:02
> > > An: Asterisk Users Mailing 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 other stdout
> output. There
> > > is some info on the voip-info wiki about using AGI.
> > >
> > > I use the phpagi 2 library, and carefully setting up the
> > > agi->verbose commmands fixes my 510 problems
> > >
> > > >
> > > > 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
> > > >
> > > >
> > > >
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> > >
> > >
> > >
> > >
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AW: [Asterisk-Users] AGI Problem
I have the phpagi 2 library too. So what did you change in details there to mute the vebrose things? > -Ursprüngliche Nachricht- > Von: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Im Auftrag von Obelix > Gesendet: Montag, 17. Oktober 2005 11:02 > An: Asterisk Users Mailing 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 other stdout > output. There is some info on the voip-info wiki about using AGI. > > I use the phpagi 2 library, and carefully setting up the > agi->verbose commmands fixes my 510 problems > > > > > 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 > > > > > > > > ___ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > This message was sent using IMP, the Internet Messaging Program. > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Variable problem
The ("/var/lib/asterisk/agi-bin/phpagi.php") is the newest form the site
updated today, and i wrote the script like other examples and i can't find a
syntax mistake inside extension.conf and the php script .(
On Thu, 13 Oct 2005 09:32:14 -0500
Moises Silva <[EMAIL PROTECTED]> 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 10/13/05, René Enskat [Teamware GmbH] <[EMAIL PROTECTED]> wrote:
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 $MSN should be in the SetCIDNum.
#!/usr/bin/php -q
get_variable("SIPUSER");
if ($ID['result'] == 0) {
$agi->verbose("SIPUSER not set -- nothing to do");
exit(1);
}
$agi->set_variable("MSN", exec("/var/lib/asterisk/agi-bin/msn4sip 111
222 333 " .$ID['data']));
?>
Output from asterisk:
-- Executing SetVar("SIP/31-79e2", "SIPUSER=31") in new stack
-- Executing AGI("SIP/31-79e2", "msn4sip.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agi
msn4sip.agi: Arrayn(n [code] => 510n [result] => n [data] =>
Invalid or unknown commandn)n
msn4sip.agi: SIPUSER not set -- nothing to do
-- AGI Script msn4sip.agi completed, returning 0
-- Executing SetLanguage("SIP/31-79e2", "de") in new stack
-- Executing SetCIDNum("SIP/31-79e2", "") in new stack
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[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 SetCIDName("SIP/31-752a", "4989427") in new stack
-- Executing SetCallerID("SIP/31-752a", ""4989427"
<4989427>") in new stack
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[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 $MSN should be in the SetCIDNum.
#!/usr/bin/php -q
get_variable("SIPUSER");
if ($ID['result'] == 0) {
$agi->verbose("SIPUSER not set -- nothing to do");
exit(1);
}
$agi->set_variable("MSN", exec("/var/lib/asterisk/agi-bin/msn4sip 111
222 333 " .$ID['data']));
?>
Output from asterisk:
-- Executing SetVar("SIP/31-79e2", "SIPUSER=31") in new stack
-- Executing AGI("SIP/31-79e2", "msn4sip.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agi
msn4sip.agi: Arrayn(n[code] => 510n[result] => n[data] =>
Invalid or unknown commandn)n
msn4sip.agi: SIPUSER not set -- nothing to do
-- AGI Script msn4sip.agi completed, returning 0
-- Executing SetLanguage("SIP/31-79e2", "de") in new stack
-- Executing SetCIDNum("SIP/31-79e2", "") in new stack
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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 Alex Lake > Gesendet: Dienstag, 4. Oktober 2005 11:47 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [Asterisk-Users] Asterisk and NAT > > You've not said much about your firewall setup. I presume > you've opened up 5060 and RTP ports? > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 phone which is behind a nat router i cant register. When the * is in official net all is working! Regards Rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
