Something wrong with my Asterisk 1.6... don't know why it cannot locate all the config files although correct path already mentioned in asterisk.conf, I use "make DESTDIR=/opt install" but the binary still look for the default paths:
Could not load features.conf
No 'modules.conf' found, no modules will be loaded.
Unable to open Asterisk database '/var/lib/asterisk/astdb': No such file or directory
No 'modules.conf' found, no modules will be loaded.
[quote]md5sum is in my system when I install the GNU coreutils.
[size=2][color=#999999]Sam 發表於 2010-9-18 00:28[/color] [url=http://www.telecom-cafe.com/telecomcafe/redirect.php?goto=findpost&pid=8957&ptid=2928][img]http://www.telecom-cafe.com/telecomcafe/images/common/back.gif[/img][/url][/size][/quote]
After installing GNU coreutils, I re-make again. So far, it keeps on compiling without errors. Let us see what are the end results later.
YH
Oh, made a stupid mistake by forgot to uncomment [directories] in asterisk.conf,... My 1.6.2 in arm linux is up and running fine now!
glad to hear you make it work.
I still have problem in compiling the Asterisk 1.4.36 source codes with the following errors:[code][/opt/usr/src/asterisk-1.4.36] # make
CC="cc" CXX="" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts
make[1]: Entering directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
make[1]: `makeopts' is up to date.
make[1]: Leaving directory `/share/HDA_DATA/.qpkg/Optware/usr/src/asterisk-1.4.36/menuselect'
In file included from chared.h:136,
from el.h:101,
from common.c:51,
from editline.c:4:
fcns.h:56:1: warning: "em_upper_case" redefined
fcns.h:50:1: warning: this is the location of the previous definition
In file included from editline.c:4:
common.c:73: error: expected identifier or '(' before numeric constant
common.c:302: error: expected identifier or '(' before numeric constant
common.c:353: error: expected identifier or '(' before numeric constant
common.c: In function 'ed_quoted_insert':
common.c:387: error: called object '8' is not a function
common.c: At top level:
common.c:397: error: expected identifier or '(' before numeric constant
common.c:441: error: expected identifier or '(' before numeric constant
common.c:466: error: expected identifier or '(' before numeric constant
common.c:485: error: expected identifier or '(' before numeric constant
common.c:498: error: expected identifier or '(' before numeric constant
common.c:524: error: expected identifier or '(' before numeric constant
common.c:537: error: expected identifier or '(' before numeric constant
In file included from editline.c:5:
emacs.c:122: error: expected identifier or '(' before numeric constant
emacs.c:288: error: expected identifier or '(' before numeric constant
emacs.c:368: error: expected identifier or '(' before numeric constant
emacs.c:382: error: expected identifier or '(' before numeric constant
emacs.c:416: error: expected identifier or '(' before numeric constant
emacs.c:470: error: expected identifier or '(' before numeric constant
emacs.c:483: error: expected identifier or '(' before numeric constant
In file included from editline.c:6:
[/code]Does anymore who have come across with it when compiling your codes under a QNAP TS-119 platform.
YH
For those who want to upgrade from Asterisk 1.4 to 1.6 please be very careful, 1.4 configuration will not work directly under 1.6 and many of the connect working well under 1.4 will be broken in 1.6, I am still having a hard time to flight with them....
In fact, under the Optware platform, there has Asterisk-1.6.2.12 ipkg install package. You do not need to compile it by yourself.
YH
You are right, I forgot to add the new Optware feeds to ipkg so I don't know 1.6 is already available, thanks!
In fact, I wanna compile the Asterisk tar ball by myself. However there are many errors coming out. I do not know how to fix it because I do not know the correct environment and libraries needed. As a result, I have to give up. I have no choice to switch of ipkg asterisk16 I want to.
YH
Maybe you can update to the latest C Dev Environment... I just figure out I can't use the latest Asterisk 1.6 ipk distribution because it is not compatible with my old glibc
Could you elaborate more about the way to update the latest C Development environment?
YH
Such as the C/C++ compiler, I've seen the above "error: expected identifier or '(' before numeric constant " usually reported on GCC 3.x.x while 4.1.1 should no longer complaint.
The one that I have has already been gcc version 4.2.3. :L
YH
The Asterisk Development Team is proud to announce the release of Asterisk
1.8.0. This release is available for immediate download at
[url]http://downloads.asterisk.org/pub/telephony/asterisk/[/url]
Asterisk 1.8 is the next major release series of Asterisk. It will be a Long
Term Support (LTS) release, similar to Asterisk 1.4. For more information about
support time lines for Asterisk releases, see the Asterisk versions page.
[url]http://www.asterisk.org/asterisk-versions[/url]
The release of Asterisk 1.8.0 would not have been possible without the support
and contributions of the community. Since Asterisk 1.6.2, we've had over 500
reporters, more than 300 testers and greater than 200 developers contributed to
this release.
You can find a summary of the work involved with the 1.8.0 release in the
sumary:
[url]http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt[/url]
A short list of available features includes:
* Secure RTP
* IPv6 Support in the SIP channel driver
* Connected Party Identification Support
* Calendaring Integration
* A new call logging system, Channel Event Logging (CEL)
* Distributed Device State using Jabber/XMPP PubSub
* Call Completion Supplementary Services support
* Advice of Charge support
* Much, much more!
A full list of new features can be found in the CHANGES file.
[url]http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup[/url]
For a full list of changes in the current release candidate, please see the
ChangeLog:
[url]http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0[/url]
Thank you for your continued support of Asterisk!
Thanks CK for letting us know the most updated information. When I have time, I may set up one server for Asterisk 1.8 for evaluation.
YH
YH,
When also have time, I will also do a compilation of asterisk 1.8. However, my problem right now is the slow processing speed for the VM image under my Windows XP.
Asterisk 1.4.38, 1.6.2.15 & 1.8.1 Now Available
The release of Asterisk 1.4.38 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Fix a crash in res_jabber by ensuring that we don't alter memory after it's
freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
The release of Asterisk 1.6.2.15 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* When using chan_skinny, don't crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA)
* Add ability for Asterisk to try both the encoded and unencoded subscription
URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman)
* Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder)
* Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Multiple fixes related to Local channels.
The release of Asterisk 1.8.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Fix issue when using directmedia. Asterisk needs to limit the codecs offered
to just the ones that both sides recognize, otherwise they may end up sending
audio that the other side doesn't understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11)
* Resolve issue where Party A in an analog 3-way call would continue to hear
ringback after party C answers.
(Patched by rmudgett)
* Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler)
* Fix problem with qualify option packets for realtime peers never stopping.
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by
jpeeler)
* Fix issue where it is possible to crash Asterisk by feeding the curl engine
invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
The Asterisk Development Team has announced the release of Asterisk 1.8.2. This
release is available for immediate download at
[url]http://downloads.asterisk.org/pub/telephony/asterisk/[/url]
The release of Asterisk 1.8.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* 'sip notify clear-mwi' needs terminating CRLF.
(Closes issue #18275. Reported, patched by klaus3000)
* Patch for deadlock from ordering issue between channel/queue locks in
app_queue (set_queue_variables).
(Closes issue #18031. Reported by rain. Patched by bbryant)
* Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
by russellb)
* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)
* Fix reloading of peer when a user is requested. Prevent peer reloading from
causing multiple MWI subscriptions to be created when using realtime.
(Closes issue #18342. Reported, patched by nivek.)
* Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0
so res_jabber doesn't think there is already an XMPP connection sending
device state. Also clean up CLI commands a bit.
(Closes issue #18272. Reported by klaus3000. Patched by Marquis42)
* Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of
setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares)
* Fixes issue with outbound google voice calls not working. Thanks to az1234
and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
For a full list of changes in this release, please see the ChangeLog:
[url]http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2[/url]
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.6.2.16.
This release is available for immediate download at
[url]http://downloads.asterisk.org/pub/telephony/asterisk/[/url]
The release of Asterisk 1.6.2.16 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested
by russellb)
* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)
* Linux and *BSD disagree on the elements within the ucred structure. Detect
which one is in use on the system.
(Closes issue #18384. Reported, patched, tested by bjm, tilghman)
* app_followme: Don't create a Local channel if the target extension does not
exist.
(Closes issue #18126. Reported, patched by junky)
* Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686.
Tested by cmbaker82)
* Resolve issue where REGISTER request with a Call-ID matching an existing
transaction is received it was possible that the REGISTER request would
overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release, please see the ChangeLog:
[url]http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.16[/url]
Thank you for your continued support of Asterisk!
The Asterisk Development Team has announced the release of Asterisk 1.4.39. This
release is available for immediate download at
[url]http://downloads.asterisk.org/pub/telephony/asterisk/[/url]
The release of Asterisk 1.4.39 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
* Resolve issue where channel redirect function (CLI or AMI) hangs up the call
instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos)
* Fix bugs in saying numbers using the Swedish language syntax
(Closes issue #18355. Reported, patched by oej)
* Fix not stopping MOH when transfered local channel queue member is answered.
The problem here is only present when local channels are used with the MOH
passthru option as well as no optimization (/nm).
Patched by jpeeler.
* Improve handling of REGISTER requests with multiple contact headers.
Patched by jpeeler.
* app_followme: Don't create a Local channel if the target extension does not
exist.
(Closes issue #18126. Reported, patched by junky)
* Revert code that changed SSRC for DTMF.
(Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686.
Tested by cmbaker82)
* Resolve issue where REGISTER request with a Call-ID matching an existing
transaction is received it was possible that the REGISTER request would
overwrite the initreq of the private structure.
(Closes issue #18051. Reported by eeman. Patched, tested by twilson)
For a full list of changes in this release, please see the ChangeLog:
[url]http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.39[/url]
Thank you for your continued support of Asterisk!
Thanks for the information. Hope this latest release can really resolve the bugs as they claim.
[b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=10959&ptid=2928]49#[/url] [i]bubblestar[/i] [/b]
On the way to compile. Will see if any difference found:victory:
[b]回復 [url=http://www.telecom-cafe.com/forum/redirect.php?goto=findpost&pid=10963&ptid=2928]50#[/url] [i]ckleea[/i] [/b]
May I know the result?
YH
So far so good. Those I want are there and working
It seems Asteriskf 1.8.2 becomes more and more popular. Have you installed the analogue card with the Asterisk 1.8.2 system?
YH
Not. Only Bubblestar C-hing has this. I want to do so but may try the USB FXO instead.
Yes. Asterisk 1.8.2 has improved some of the bugs in IAX that I experienced before. Others are remaining stable so far.
Are all of you using YUM to upgrade or yourcompile the source code yourself?
YH
WE compile from source code.
Mine is like this[code]svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
cd 1.8
./configure
#this is ony for format MP3 - SVN required
contrib/scripts/get_mp3_source.sh
contrib/scripts/get_ilbc_source.sh
make menuselect
make
[/code]
Seems YUM to upgrade to Asterisk 1.8.2 is not ready yet.
I compiled.
The Asterisk Development Team has announced security releases for Asterisk
branches 1.4, 1.6.1, 1.6.2, and 1.8. The available security releases are
released as versions 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4.
These releases are available for immediate download at
[url]http://downloads.asterisk.org/pub/telephony/asterisk/releases[/url]
The releases of Asterisk 1.4.39.2, 1.6.1.22, 1.6.2.16.2, and 1.8.2.4 resolve an
issue that when decoding UDPTL packets, multiple stack and heap based arrays can
be made to overflow by specially crafted packets. Systems configured for
T.38 pass through or termination are vulnerable. The issue and resolution are
described in the AST-2011-002 security advisory.
For more information about the details of this vulnerability, please read the
security advisory AST-2011-002, which was released at the same time as this
announcement.
For a full list of changes in the current release, please see the ChangeLog:
[url]http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.4.39.2[/url]
[url]http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.1.22[/url]
[url]http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6.2.16.2[/url]
[url]http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.2.4[/url]
Security advisory AST-2011-002 is available at:
[url]http://downloads.asterisk.org/pub/security/AST-2011-002.pdf[/url]
Thank you for your continued support of Asterisk!
Upgraded last night. Thanks