[Asterisk-Dev] possible bug in app_dial.c ?
Hi,
i am not sure but this snippet of code in app_dial.c 1.188 looks
suspicious:
1.128 markster 432: while (o) {
..
539: }
1.84 markster 540: /* Hangup the original channel now, in cas
e we needed it */
541: ast_hangup(winner);
1.24 markster 542: continue;
1.22 markster 543: }
1.7 markster 544: f = ast_read(winner);
545: if (f) {
..
654: }
655: o = o-next;
656: }
i wonder if the continue in line 542 shouldn't be preceded by o = o-next;
or better yet change the while() into a proper for loop
432: for (o=outgoing; o; o = o-next) {
Comments anyone ?
cheers
luigi
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Re: [Asterisk-Dev] Problem with hanging up a SIP channel
On Thursday 24 November 2005 20:15, imran ahmed wrote: Just a note that you cannot hangup the original incoming channel using ast_hangup (use ast_softhangup only on that channel) because asterisk expects that channel to be returned back to the dialplan. Yes, I suspected that already as my tests showed ast_softhangup works on the original channel. But thanks for the hint anyway :-) C'ya, Marc -- Marc Haisenko Comdasys AG Rüdesheimer Straße 7 D-80686 München Tel: +49 (0)89 - 548 43 33 0 Fax: +49 (0)89 - 548 43 33 29 e-mail: [EMAIL PROTECTED] http://www.comdasys.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Tr: RE : [Asterisk-Users] Asterisk doesn't start
Remarque : message transféré en pièce jointe. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Yes, beta2 works perfectly, but 1.2 released version gives me this error. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : vendredi 25 novembre 2005 11:24 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] Asterisk doesn't start Hello, You built asterisk on freebsd ? Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello Whan starting astersik(1.2) (asterisk -vvc), I get this message : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined s ymbol ast_config_load What did I forgot to do? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ---End Message--- ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] asterisk 1.2 g729 compile errors
Hello Ali,
try this patch against the vm_types_linux32.h include. Worked for me.
Kind regards,
Christian Braun.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Diyanat Ali
Sent: Thursday, November 24, 2005 6:33 PM
To: asterisk-dev@lists.digium.com
Subject: [Asterisk-Dev] asterisk 1.2 g729 compile errors
I had sucessfully compiled and tested free g729 codec for asterisk 1.09
based on Intel ipp with patch and instructions from
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
I am trying to compile again with asterisk 1.2 and getting the following
errors
vm/include/sys/vm_types_linux32.h:36: syntax error before
use_ast_cond_t_instead_of_pthread_cond_t
vm/include/sys/vm_types_linux32.h:36: warning: no semicolon at end of struct
or union
vm/include/sys/vm_types_linux32.h:37: warning: data definition has no type
or storage class
vm/include/sys/vm_types_linux32.h:40: syntax error before '}' token
vm/include/sys/vm_types_linux32.h:40: warning: data definition has no type
or storage class
vm/include/sys/vm_types_linux32.h:51: syntax error before
use_ast_mutex_t_instead_of_pthread_mutex_t
vm/include/sys/vm_types_linux32.h:51: warning: no semicolon at end of struct
or union
vm/include/sys/vm_types_linux32.h:53: syntax error before '}' token
vm/include/sys/vm_types_linux32.h:53: warning: data definition has no type
or storage class
vm/include/sys/vm_types_linux32.h:57: syntax error before
use_ast_cond_t_instead_of_pthread_cond_t
vm/include/sys/vm_types_linux32.h:57: warning: no semicolon at end of struct
or union
vm/include/sys/vm_types_linux32.h:58: warning: data definition has no type
or storage class
vm/include/sys/vm_types_linux32.h:60: syntax error before '}' token
vm/include/sys/vm_types_linux32.h:60: warning: data definition has no type
or storage class
In file included from samples/decoder.h:27,
from samples/codec_g729.c:40:
samples/util_d.h:39:1: warning: BITSTREAM_FILE_FRAME_SIZE redefined
In file included from samples/encoder.h:27,
from samples/codec_g729.c:39:
samples/util_e.h:39:1: warning: this is the location of the previous
definition
samples/codec_g729.c: In function `g729tolin_new':
samples/codec_g729.c:168: warning: assignment from incompatible pointer type
samples/codec_g729.c: In function `g729tolin_framein':
samples/codec_g729.c:244: warning: assignment from incompatible pointer type
samples/codec_g729.c: In function `lintog729_frameout':
samples/codec_g729.c:285: warning: assignment from incompatible pointer type
make: *** [samples/codec_g729.o] Error 1
I guess the patch needs to be updated to work with asterisk 1.2
Diyanat
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begin 666 vm_types_linux32.h.diff
M*BHJ('9M7W1Y5S7VQI;G5X,S(N:YOFEG5=E9!.;W8@,C,@,[EMAIL PROTECTED],#4Z
M,#$@,C [EMAIL PROTECTED]'EP97-?;[EMAIL PROTECTED],BYH5=E9!.;W8@,C,@,[EMAIL
PROTECTED]
M,#8Z,34@,C P-0HJ*BHJ*BHJ*BHJ*BHJ*BH**BHJ(#,S+#0P(HJ*BH*( *
M( O*B!V;5]E=F5N=YH(HOB @='EP961E9B!S=')U8W0@PHA( EP=AR
M96%D7V-O;F1?=!C;VYD.PHA( EP=AR96%D7VUU=5X7W0@;[EMAIL PROTECTED]B @
M6EN=!M86YU86P[B @6EN=!S=%T93L*(!]('9M7V5V96YT.PHM+2T@
M,S,L-# @+2TM+0H@( H@(\J('9M7V5V96YT+F@@*B\*(!T7!E95F('-T
MG5C=![B$@6%S=%]C;[EMAIL PROTECTED](2 )87-T7VUU=5X7W0@;75T
[EMAIL PROTECTED]B @6EN=!M86YU86P[B @6EN=!S=%T93L*(!]('9M7V5V96YT
M.PHJ*BHJ*BHJ*BHJ*BHJ*BH**BHJ(#0X+#8Q(HJ*BH*( *( O*B!V;5]M
M=71EYH(HOB @='EP961E9B!S=')U8W0@PHA( EP=AR96%D7VUU=5X
M7W0@:%N9QE.PH@( EI;G0@:7-?=F%L:60[B @?2!V;5]M=71E#L*( *
M( O*B!V;5]S96UAAOF4N: J+PH@('1Y5D968@W1R=6-T('L*(2 @
M( @'1HF5A9%]C;[EMAIL PROTECTED](2 @( @'1HF5A9%]M=71E%]T
M(UU=5X.PH@( @(!I;[EMAIL PROTECTED];G0[B @?2!V;5]S96UAAOF4[B @
MBTM+2 T.PV,2 M+2TMB @B @[EMAIL PROTECTED];[EMAIL PROTECTED]: J+PH@('1Y5D
M968@W1R=6-T('L*(2 )87-T7VUU=5X7W0@:%N9QE.PH@( EI;G0@:7-?
M=F%L:60[B @?2!V;5]M=71E#L*( *( O*B!V;5]S96UAAOF4N: J
M+PH@('1Y5D968@W1R=6-T('L*(2 @( @87-T7V-O;F1?=!C;VYD.PHA
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1=FU?V5M87!H;W)E.PH@( H`
`
end
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[Asterisk-Dev] Help Connecting Cisco Router to Asterisk
I know some one has done this, I need help connecting Cicso Router MC3800 Series with FXO port to Asterisk. I have got the basic config but I dont get a dial tone. Some one please help. Thanks Diseyi ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] asterisk 1.2 g729 compile errors
Hello! Braun I was able to compile and test the codecs after applying the patch to vm_types_linux32.h Thanks Diyanat From: [EMAIL PROTECTED] Reply-To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com To: 'Asterisk Developers Mailing List' asterisk-dev@lists.digium.com Subject: RE: [Asterisk-Dev] asterisk 1.2 g729 compile errors Date: Fri, 25 Nov 2005 14:46:59 +0100 Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 25 Nov 2005 13:50:37.0076 (UTC) FILETIME=[36D6CD40:01C5F1C7] Hello Ali, try this patch against the vm_types_linux32.h include. Worked for me. Kind regards, Christian Braun. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diyanat Ali Sent: Thursday, November 24, 2005 6:33 PM To: asterisk-dev@lists.digium.com Subject: [Asterisk-Dev] asterisk 1.2 g729 compile errors I had sucessfully compiled and tested free g729 codec for asterisk 1.09 based on Intel ipp with patch and instructions from http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ I am trying to compile again with asterisk 1.2 and getting the following errors vm/include/sys/vm_types_linux32.h:36: syntax error before use_ast_cond_t_instead_of_pthread_cond_t vm/include/sys/vm_types_linux32.h:36: warning: no semicolon at end of struct or union vm/include/sys/vm_types_linux32.h:37: warning: data definition has no type or storage class vm/include/sys/vm_types_linux32.h:40: syntax error before '}' token vm/include/sys/vm_types_linux32.h:40: warning: data definition has no type or storage class vm/include/sys/vm_types_linux32.h:51: syntax error before use_ast_mutex_t_instead_of_pthread_mutex_t vm/include/sys/vm_types_linux32.h:51: warning: no semicolon at end of struct or union vm/include/sys/vm_types_linux32.h:53: syntax error before '}' token vm/include/sys/vm_types_linux32.h:53: warning: data definition has no type or storage class vm/include/sys/vm_types_linux32.h:57: syntax error before use_ast_cond_t_instead_of_pthread_cond_t vm/include/sys/vm_types_linux32.h:57: warning: no semicolon at end of struct or union vm/include/sys/vm_types_linux32.h:58: warning: data definition has no type or storage class vm/include/sys/vm_types_linux32.h:60: syntax error before '}' token vm/include/sys/vm_types_linux32.h:60: warning: data definition has no type or storage class In file included from samples/decoder.h:27, from samples/codec_g729.c:40: samples/util_d.h:39:1: warning: BITSTREAM_FILE_FRAME_SIZE redefined In file included from samples/encoder.h:27, from samples/codec_g729.c:39: samples/util_e.h:39:1: warning: this is the location of the previous definition samples/codec_g729.c: In function `g729tolin_new': samples/codec_g729.c:168: warning: assignment from incompatible pointer type samples/codec_g729.c: In function `g729tolin_framein': samples/codec_g729.c:244: warning: assignment from incompatible pointer type samples/codec_g729.c: In function `lintog729_frameout': samples/codec_g729.c:285: warning: assignment from incompatible pointer type make: *** [samples/codec_g729.o] Error 1 I guess the patch needs to be updated to work with asterisk 1.2 Diyanat ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev vm_types_linux32.h.diff ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] dialplan hint case sensitive
I noticed that the hint was not working when I write the channel name
upcase for zap channel
for example
exten = 1,hint,zap/1
does not work
exten = 1,hint,ZAP/1
does not work
exten = 1,hint,Zap/1
work because the zt_requst create the channel with this name.
Every channel type match in the code is done with strcasecmp except in
the ast_hint_state_changed and this could a problem
so this is the patch for the CVS v1-2 but it could be manually applied
to all asterisk versions
Index: asterisk/pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.303
diff -u -r1.303 pbx.c
--- asterisk/pbx.c14 Nov 2005 19:00:38 -1.303
+++ asterisk/pbx.c25 Nov 2005 14:50:36 -
@@ -1888,7 +1888,7 @@
ast_copy_string(buf, ast_get_extension_app(hint-exten),
sizeof(buf));
parse = buf;
for (cur = strsep(parse, ); cur; cur = strsep(parse, )) {
-if (strcmp(cur, device))
+if (strcasecmp(cur, device))
continue;
/* Get device state for this hint */
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Re: [Asterisk-Dev] MeetMe assumes 20ms...
On Thu, 24 Nov 2005, Dan Austin wrote: The change from careful_write to write between version 1.2.0-beta1 and 1.2.0 did not take into consideration that a caller might be using a packetization of more than 20ms. I've been working on making app_meetme handle different size audio frames, and have had some success. The last obsticle is setting up the read buffer. My 'C' is quite rusty, and I cannot figure out a way to determine the size of the read buffer. char __buf[CONF_SIZE + AST_FRIENDLY_OFFSET]; char *buf = __buf + AST_FRIENDLY_OFFSET; What I hoped to do is determine the size of buf, and if needed use realloc to grow/shrink the read buffer to the proper size. Any hints would be appreciated, even a 'take another class in C' This is also being discussed on the Bugtracker at: http://bugs.digium.com/view.php?id=5697 -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number
Muhammad Asim Sajjad wrote: Hi, i am very thankfull to you on this helping material but i require more help in this regard. My problem is that i want to know how can i detect that the Human of Answering machine is connected with my called phone number. BackgroundDetect or MachineDetect - really a *-users question though. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Require Help in Detection of Human Voice or Answering machine on Called telephone number
Matt Riddell wrote: Muhammad Asim Sajjad wrote: Hi, i am very thankfull to you on this helping material but i require more help in this regard. My problem is that i want to know how can i detect that the Human of Answering machine is connected with my called phone number. BackgroundDetect or MachineDetect - really a *-users question though. BackgroundDetect plays a message while listening for DTMF, so that's not what he wants. MachineDetect waits for silence, but doesn't really differentiate between live and recorded voice, so that's not what he wants. Differentiating between live and recorded voice is a very hard problem, possibly impractical. Nobody has yet succeeded in producing a meaningful version of such a function. This is most definitely a developer issue, but probably not one that will result in an effective solution. Regards, Steve ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
