Re: [Asterisk-Users] Asterisk 0.7.0
Any link where to directly find the main differences between 0.5.0.
and
0.7.0??
from ChangeLog off CVS
Asterisk 0.7.0
-- Removed MP3 format and codec
-- Can now load and unload SIP,IAX,IAX2,H323 channels without core
-- Fixed various compiler warnings and clean up source tree
-- Preliminary AES Support
-- Fix SIP REINVITE
-- Outbound SIP registration behind NAT using externip
-- More CLI documentation and clean up
-- Pin numbers on MeeMe
-- Dynamic MeetMe conferences are more consistent with static conferences
-- Added channel variables ${HANGUPCAUSE}, ${SIPDOMAIN}, ${TIMESTAMP},
${ACCONTCODE}
-- ODBC support for logging CDRs
-- Indications for Norway and New Zeland
-- Major redesign of app_voicemail
-- Syslog support
-- Reload logfiles with CLI command 'logger reload' and rotate logs with
logger rotate'
-- Configurable DEBUG, NOTICE, WARNING, ERROR and ast_verbose messages now
appear on remote console
-- Properly reaping any zombie processes
-- Added applications SayUnixTime, SetCDRUserField, HasNewVoicemail,
ZapScan, Random, ResetCDR, NoCDR
-- Make PRI Hangup Cause available to the dialplan
-- Verify included contexts in extensions.conf
-- Add DESTDIR support for building RPMs and packages
-- Do route lookups on OpenBSD
-- Add support for building on FreeBSD and OS X
-- Add support for PostgreSQL in Voicemail
-- Translate SIP hangup cause to PRI hangup cause where needed
-- Better support for MOH in IAX2
-- Fix SIP problem where channels were not removed on BYE
-- Display codecs by name
-- Remove MySQL and put PGSql instead for licensing reasons
-- Better capability matching in SIP
-- Full IBR4 compliance for chan_zap
-- More flexible CDR handling
-- Distinguish between BUSY and FAILURE on outbound calls
-- Add initial support for SCCP via chan_skinny
-- Better support for Future Group B signaling
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RE: [Asterisk-Users] Asterisk 0.7.0
Yep that was the problem that was fixed! :) bkw On Tue, 13 Jan 2004, Dan Austin wrote: Would it be imprudent to ask what is broke with chan_h323? I finally managed to get it compiled using 0.7.0, and can get calls in and out of * with it, but limited testing shows no working RTP, and h.323 debug shows that * thinks the local address is 127.0.0.1. If that is the concern with chan_h323 in 0.7.0, I'll happily wait for the next release. Otherwise I guess I have more digging to do. Dan -Original Message- From: Manuel João S. Costa Amaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 11:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk 0.7.0 A Ter, 2004-01-13 às 16:01, Brian West escreveu: Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) I've tryied it today, for a few minutes. I think there's problem with oh323 also. If the problem persists, tomorrow i report. Anyone ? Cya -- Joao Amaro ([EMAIL PROTECTED]) Braga, Portugal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
Better. Now it is one-way audio. The call is being directed to a Cisco
Call Manager, and the call looks good from the h.323 debug except the
sending stream:
-- Started logical channel: sending G.711-uLaw-64k{sw}
-- channelsOpen = 1
-- remoteIpAddress: 0.0.0.0
-- remotePort: 0
-- ExternalIpAddress: 134.122.5.125
-- ExternalPort: 11646
I've seen some comments that chan_h323 might have this problem with
calls to a Call Manager, but the threads seemed to just fade away,
with the only recommendation to use chan_oh323.
Any thoughts on where I can chase a solution?
Thanks,
Dan
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 10:06 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk 0.7.0
Yep that was the problem that was fixed! :)
bkw
On Tue, 13 Jan 2004, Dan Austin wrote:
Would it be imprudent to ask what is broke with chan_h323?
I finally managed to get it compiled using 0.7.0, and can
get calls in and out of * with it, but limited testing shows no
working RTP, and h.323 debug shows that * thinks the local address is
127.0.0.1.
If that is the concern with chan_h323 in 0.7.0, I'll happily wait for
the next release. Otherwise I guess I have more digging to do.
Dan
-Original Message-
From: Manuel João S. Costa Amaro [mailto:[EMAIL PROTECTED]
Sent: Tuesday, January 13, 2004 11:38 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk 0.7.0
A Ter, 2004-01-13 às 16:01, Brian West escreveu:
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
Give it a few days and I bet we will. because chan_h323 is broken
also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing
it.)
I've tryied it today, for a few minutes. I think there's problem with
oh323 also.
If the problem persists, tomorrow i report.
Anyone ?
Cya
--
Joao Amaro ([EMAIL PROTECTED])
Braga, Portugal
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Re: [Asterisk-Users] Asterisk 0.7.0
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
GotoIfTime application. I'll own up to writing the broken code. The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c 11 Jan 2004 09:19:16 - 1.92
+++ pbx.c 13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
Re: [Asterisk-Users] Asterisk 0.7.0
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and the
GotoIfTime application. I'll own up to writing the broken code. The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c 11 Jan 2004 09:19:16 - 1.92
+++ pbx.c 13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is critical?
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[Asterisk-Users] Asterisk 0.7.0
Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! Mark p.s. there was no 0.6.0 release. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
On Tuesday 13 January 2004 02:27, WipeOut wrote: Tilghman Lesher wrote: On Tuesday 13 January 2004 00:10, Mark Spencer wrote: Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! There is one bug so far and it's critical. It breaks includes and the GotoIfTime application. I'll own up to writing the broken code. The fix is very simple, though (attached). Why not quickly patch the source an release 0.7.1 if the bug is critical? We're planning to do that, but there's going to be a lag between planning a release and getting a release out. For people who want to use 0.7.0 right away, it's better to release news of the discovery of the bug right away, not wait for a new release. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
I think the exchange below shows us that before 0.8.0 comes
out, maybe there should be a 0.8.0-beta then after no problems are
reported in a few week period a 0.8.0-release candidate and
ten 0.8.0 itself.
It's hard to call a realease stable until a number of people outside
the developer's lab have used it for a while.
The other idea is that everyone just knows that x.y.0 == beta
and they all wait for a .1 or .2 realease.
--- WipeOut [EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all
the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and
the
GotoIfTime application. I'll own up to writing the broken code.
The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c11 Jan 2004 09:19:16 - 1.92
+++ pbx.c13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
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Re: [Asterisk-Users] Asterisk 0.7.0
Brian West wrote: Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) Maybe an idea would be to run an RC (Release Candidate) system so that stable releases are tested for at least a day or two before being declared stable and avoid the 3am rush.. :) Not to take anything away from the awesome job you guys are doing by having a stable branch and all.. it may just make it a little less of a rush and a little less stressfull.. Just a thought.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
- Original Message - From: Chris Albertson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 12:11 PM Subject: Re: [Asterisk-Users] Asterisk 0.7.0 I think the exchange below shows us that before 0.8.0 comes out, maybe there should be a 0.8.0-beta then after no problems are reported in a few week period a 0.8.0-release candidate and ten 0.8.0 itself. It's hard to call a realease stable until a number of people outside the developer's lab have used it for a while. The other idea is that everyone just knows that x.y.0 == beta and they all wait for a .1 or .2 realease. Does anyone know if there was another product that released a version x.6 or x.7 that could have started this rush for an asterisk 0.7.0?\ - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
On Tue, 2004-01-13 at 11:11, Chris Albertson wrote:
I think the exchange below shows us that before 0.8.0 comes
out, maybe there should be a 0.8.0-beta then after no problems are
reported in a few week period a 0.8.0-release candidate and
ten 0.8.0 itself.
It's hard to call a realease stable until a number of people outside
the developer's lab have used it for a while.
The other idea is that everyone just knows that x.y.0 == beta
and they all wait for a .1 or .2 realease.
Of course then there is the thought that the version number is nothing
more than name decoration. It is just a number given to make it easier
to identify a set of bugs.
--- WipeOut [EMAIL PROTECTED] wrote:
Tilghman Lesher wrote:
On Tuesday 13 January 2004 00:10, Mark Spencer wrote:
Okay, it's 15 minutes late, but it's out, thanks very much to all
the
people who worked so hard this weekend to make this possible!
There is one bug so far and it's critical. It breaks includes and
the
GotoIfTime application. I'll own up to writing the broken code.
The
fix is very simple, though (attached).
-Tilghman
Index: pbx.c
===
RCS file: /usr/cvsroot/asterisk/pbx.c,v
retrieving revision 1.92
diff -u -r1.92 pbx.c
--- pbx.c 11 Jan 2004 09:19:16 - 1.92
+++ pbx.c 13 Jan 2004 07:21:12 -
@@ -2922,7 +2922,7 @@
return;
}
-#if 0
+#if 1
s1 = s1 * 30 + s2/2;
if ((s1 0) || (s1 = 24*30)) {
ast_log(LOG_WARNING, %s isn't a valid star time. Assuming no
time.\n, times);
Why not quickly patch the source an release 0.7.1 if the bug is
critical?
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Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk 0.7.0
A Ter, 2004-01-13 às 16:01, Brian West escreveu: Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) I've tryied it today, for a few minutes. I think there's problem with oh323 also. If the problem persists, tomorrow i report. Anyone ? Cya -- Joao Amaro ([EMAIL PROTECTED]) Braga, Portugal signature.asc Description: Esta =?ISO-8859-1?Q?=E9?= uma parte de mensagem assinada digitalmente
Re: [Asterisk-Users] Asterisk 0.7.0
It's hard to call a realease stable until a number of people outside the developer's lab have used it for a while. 0.7.0 was never taged nor advertised as stable. Its just the last release before the 0.9.x branch is tagged. 0.9.x will go thru this process of release after release till the 1.0_RC's are tested and ready. 0.9.x will be the road to 1.0 stable. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
Any link where to directly find the main differences between 0.5.0. and 0.7.0?? Thanks a lot!! Sam - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 10:09 AM Subject: Re: [Asterisk-Users] Asterisk 0.7.0 On Tuesday 13 January 2004 02:27, WipeOut wrote: Tilghman Lesher wrote: On Tuesday 13 January 2004 00:10, Mark Spencer wrote: Okay, it's 15 minutes late, but it's out, thanks very much to all the people who worked so hard this weekend to make this possible! There is one bug so far and it's critical. It breaks includes and the GotoIfTime application. I'll own up to writing the broken code. The fix is very simple, though (attached). Why not quickly patch the source an release 0.7.1 if the bug is critical? We're planning to do that, but there's going to be a lag between planning a release and getting a release out. For people who want to use 0.7.0 right away, it's better to release news of the discovery of the bug right away, not wait for a new release. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
Would it be imprudent to ask what is broke with chan_h323? I finally managed to get it compiled using 0.7.0, and can get calls in and out of * with it, but limited testing shows no working RTP, and h.323 debug shows that * thinks the local address is 127.0.0.1. If that is the concern with chan_h323 in 0.7.0, I'll happily wait for the next release. Otherwise I guess I have more digging to do. Dan -Original Message- From: Manuel João S. Costa Amaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 11:38 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk 0.7.0 A Ter, 2004-01-13 às 16:01, Brian West escreveu: Why not quickly patch the source an release 0.7.1 if the bug is critical? Give it a few days and I bet we will. because chan_h323 is broken also in 0.7.0 (JerJer :P but him and I stayed up till 3 am fixing it.) I've tryied it today, for a few minutes. I think there's problem with oh323 also. If the problem persists, tomorrow i report. Anyone ? Cya -- Joao Amaro ([EMAIL PROTECTED]) Braga, Portugal ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Asterisk 0.7.0
- Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 4:23 PM Subject: Re: [Asterisk-Users] Asterisk 0.7.0 It's hard to call a realease stable until a number of people outside the developer's lab have used it for a while. 0.7.0 was never taged nor advertised as stable. Its just the last release before the 0.9.x branch is tagged. 0.9.x will go thru this process of release after release till the 1.0_RC's are tested and ready. 0.9.x will be the road to 1.0 stable. Did I miss something or did we just skip another version one day after closing the previous one? Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
