to the Digium D40?
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know it's
old but it works and does what I need. Are there differences in versions on
how the above would work?
sip.conf
[general]
externhost = foo.no-ip.org
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with the nsca client.
Works great.
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might find that helpful.
A side from that, we are currently rolling our own Ubuntu Precise
packages based on 1.8.7.1, backporting patches we require. Everything
is on github[1] if you want to do the same.
[1] https://github.com/kickstandproject/asterisk/wiki/Installing-Packages
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Paul
is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip
Just use an existing library, rather then rolling your own.
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Github
on that box,
meaning, did you also compile libpri before compiling Asterisk?
How about watching your Asterisk log files during Asterisk startup to see any
output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full)
*CLI pri show version
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, adding another vendor into the mix for use to support
does not make sense at this time.
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the SBC to do?
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the asterisk testsuite for some examples[1]. You could use a
combination of StarPy and pjsua (python bindings) to do this.
[1] http://svnview.digium.com/svn/testsuite/asterisk/trunk/
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On 12-05-31 12:28 PM, asterisk users wrote:
Connecting to downloads.asterisk.org 76.164.171.233|:80...
failed: Connection timed out.
WORKSFORSOME?
http://www.downforeveryoneorjustme.com/http://downloads.asterisk.org
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, and thanks in advance.
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(Python) has this functionality by using the Twisted. It simply
monitors events on the AMI, and if the connect break, can be setup to
re-connect.
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On 12-05-03 03:47 PM, Paul Belanger wrote:
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management
interface using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts
] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not
permitted
'Operation not permitted' indicates a permission issue. Check and make
sure asterisk has the permissions to access your eth interfaces.
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Paul Belanger
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conferencing you may need to look into
Vmukti project http://sourceforge.net/projects/vmukti/
Please explain your requirements so anyone can help you in better way.
Actually, if OP upgrades to Asterisk 10 they will get video conferencing
with app_confbridge.
-- Paul Belanger Digium, Inc. | Software
] https://reviewboard.asterisk.org/r/1764/
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On 12-03-08 04:41 AM, Jon Farmer wrote:
Hi
Just realised this is due to a FIFO blocking. Fixed that and all back to normal.
How did you fix it? Will help others playing at home (and me too).
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it. It is
likely a deadlock so attach the required information for it.
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5.7
asterisk-1.8.9.2
dahdi-linux-complete-2.6.0
libpri-1.4.12
What am I missing?
For what ever reason, asterisk is crashing. You'll need to generate a
backtrace[1].
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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twitter
-creating
the directory. I even noted this issue on reviewboard[1], however it
was never implemented.
[1] https://reviewboard.asterisk.org/r/654/#review2370
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be helpful too, please attach a debug[1] log.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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as the asterisk-dahdi package comes for the same repository as
the asterisk package you should be fine.
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box. What if OP is running more then 1 asterisk box, manually
compiling asterisk and installing it each time would not be the best
solution.
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recently fixed in 1.8.8.0+
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The only key to unlcock it when it gets locked is by restarting asterisk
I am now using 1.6.2.22. Would I then be better upgrading to 1.8 ? Does
1.8 has fewer bugs ?!
Showing us the deadlock is the best thing you can do right now.
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twitter
On 12-02-06 09:23 AM, Jonas Kellens wrote:
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
/Getting+a+Backtrace
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On 12-02-03 09:05 AM, Jonas Kellens wrote:
On 02/03/2012 02:52 PM, Paul Belanger wrote:
On 12-02-03 07:55 AM, Jonas Kellens wrote:
This is a production server. Will it affect theserver ?I already have
dont_optimize checked in the debug options.
Yes, reproduce the issue on your test
On 12-02-03 09:53 AM, Jonas Kellens wrote:
On 02/03/2012 03:48 PM, Paul Belanger wrote:
On 12-02-03 09:05 AM, Jonas Kellens wrote:
On 02/03/2012 02:52 PM, Paul Belanger wrote:
On 12-02-03 07:55 AM, Jonas Kellens wrote:
This is a production server. Will it affect theserver ?I already have
if you
can attach your astdb file from asterisk 1.8.8.0 too, it will help to
see what is happening.
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?
What OS?
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On 12-01-17 05:16 AM, Jonas Kellens wrote:
Hello list,
where can I post the output of the trace taken from a file :
/tmp/core.sip.pbx.tld-2012-01-17T11:09:13+0100
I want someone to tell me what went wrong.
Here is usually fine.
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twitter
On 12-01-05 05:12 AM, Durgesh Mishra wrote:
Hi friend,
Is asterisk 1.8 support video trascoding ?
No version of asterisk supports it.
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)
same = n,Hangup()
exten= help,1,Answer()
same = n,NoOp(you are at help section)
same = n,Hangup()
We have DTMF based tests for the testsuite[1] that you could use.
[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/
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release.
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them for this mailing list.
If we continue to receive them, I'll ask to have you removed from the list.
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On 11-12-12 11:36 AM, --[ UxBoD ]-- wrote:
Hello all,
I have recently upgraded to version 1.8.7.2 and have started to see the
following errors in the logs:
From what version?
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Check us out
to do a faux ring which xlite interprets as a
MOH request, so if you don't want to patch/recompile, just take the r off of
Dial.
Why are you manually patching asterisk? Have you created an issue in
JIRA about this?
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/lib/ruby/1.8/rake.rb:1998:in `run'
/usr/bin/rake:28
I'm completely new to ruby, rails and so on.
Suggestions ?
Contact the maintainer for support? Since this is not an asterisk issue.
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, a release announcement will be posted to
the mailing lists[1]. In the meantime you should be able to use
asterisk-10.0.0-rc2 for your needs.
[1]
http://lists.digium.com/pipermail/asterisk-announce/2011-November/thread.html
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safe_asterisk.
We should consider updating the Makefile in asterisk trunk to start
using them. More and more OS are starting to support them.
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-r -x 'database deltree example'
in /etc/init.d/asterisk or safe_asterisk?
Easier to use cli.conf
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...
Nope, nobody submitted any patches for it. So anything now would have to
be submitted into trunk, which would make Asterisk 11 the next version
to support it.
Again, assuming somebody submits a patch.
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, just modify new_chat_message()
to allocate a new ast_msg, fill in the appropriate values with the
ast_msg_set_set_() functions, and then call ast_msg_queue() on it.
Terry
What Terry said.
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Check
error)
Which version of asterisk is this?
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Greetings,
If you are planning on attending Astricon, please take the time to
attend the GPG key signing event. More information can be found on the
wiki page[1].
[1]
https://wiki.asterisk.org/wiki/display/~pabelanger/Astricon+2011+Key+signing+event
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but in the end
it would have been too much work trying to manage all the was to add
users in the different operating systems.
I had planned to add a script into contrib folder for it, but sadly
never got around to it. Perhaps I'll spend some time on it at Astricon.
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/ASTERISK-18738
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!= ERROR so you should be fine with chan_dahdi.so
As for the other modules, if you are not using them add
noload = res_config_pgsql.so
noload = res_config_ldap.so
into your modules.conf
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source. I suggest you open
an issue on the tracker. If you can determine when the issue started,
it would also help.
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,AbsoluteTimeout(60)
exten = 222,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = 222,n,Dial(SIP/${EXTEN},,KkTt)
exten = 222,n,Hangup();
could you please see this code and tell me waht is wrong
*CLI core show application Dial
Look at the 'L' flag
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On 11-09-25 01:54 AM, Антон Квашёнкин wrote:
Just use cli aliases, provided by res_clialiases.so.
2011/9/25 Bruce Bbruceb...@gmail.com
Please don't feed the trolls. Thanks.
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this is open source software everything is
listed online [1]. Which was done by mvanbaak, an asterisk community
member, not a Digium employee.
[1] http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121
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-users/2010-April/247084.html
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so defensive. No one gains
anything from changes like this. I am sure Digium can afford one afternoon
meeting to decide what the commands naming convention should be for the next
20 years.
I don't even know how to reply to this, so I won't. Thanks for all the
fish.
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Digium, Inc
information about the problem would be helpful (EG: console
output). Also, if you have not seen:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
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On 11-09-12 12:07 PM, Danny Nicholas wrote:
I think that is your best bet. 1.8.6 unless somebody has a good reason not
to.
You actually might want to test with 1.8.7.0-rc1, this will fix 2 big
issue. A performance regressions and timerfd.
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with no response
== Spawn extension (dial-fish, s, 6) exited non-zero on 'SIP/101'
This is a bug with netsock2.c unable to resolve a hostname or SIP peer
into an IP address.
https://issues.asterisk.org/jira/browse/ASTERISK-17146
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:
exten = _*7XXX,1,Pickup(${EXTEN:2}@PICKUPMARK)
exten = _*7XXX,n,Hangup()
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On 11-09-01 07:04 AM, Tim King wrote:
I have found numerous claims that 1.8 can do T.38 gateway with a patch,
however I am yet to find the patch or any instructions on implementing it.
Anyone have a link?
Asterisk-10.0.0-beta1 is another option.
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). I don't think VirtualBox is up
to real-time stuff.
What timing module do you use? I recall on several cases that the
pthreads timing module worked better than the timerfd one.
1.8.7.0-rc1 should have a few fixes for timerfd. It would be good to
get some feedback from testers.
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-beta1 is another option.
I've been testing the T.38 functionality in 10.0.0-beta1 with very successful
results.
Any information about the results you can post is good. I know we are
interested in seeing the results.
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On 11-08-30 05:16 AM, Kolmisoft Marketing wrote:
NOTE: This is not attempt to sell you anything. No product or service is
sold/marketed in the video.
That maybe the case, but this is still a non-commercial mailing list.
Please use asterisk-biz in the future.
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Digium, Inc
had the same problem and had some solutions?
Asterisk 1.4 does not have support SIP over TCP. It was added in
Asterisk 1.6.0.
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patches... etc.
Well, both 1.4 and 1.6 branches are unsupported so you should move to
asterisk 1.8 and test.
There is no 'trick' for adding TCP over SIP into Asterisk 1.4, that is
not realistic.
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somebody else on the -users list can comment.
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On 11-08-21 10:36 AM, Paul Belanger wrote:
On 11-08-21 02:54 AM, Jeremy Kister wrote:
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
On 11-08-21 02:54 AM, Jeremy Kister wrote:
On 8/20/2011 12:46 PM, Paul Belanger wrote:
Anybody able to confirm the patch in ASTERISK-18301[1] fixes outbound
google voice calls?
confirmed on asterisk 1.8.6.0-rc1
pre-patch behavior: ring-no-answer
post-patch behavior: expected
Thanks
is enabled, I'll bump and rebuild the lucid package with
PRI support.
[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Addingtheproposedbranch%28Optional%29
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.
But to answer your question, there is no way to change SIP response
codes via the dialplan, you would have to modify the source for
chan_sip.c (something I do not recommend).
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?
I was hoping I could just change the following line:-
if ((x 63) || (x 0)) {
https://issues.asterisk.org/jira/browse/ASTERISK-15463
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On 11-08-03 02:01 AM, Richard Zulu wrote:
I have used gdb so that I can perform a backtrace however the program
executes and exits without a stack thus not helpful.
Any help is appreciated!
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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Paul Belanger
Digium, Inc
On 11-08-03 07:36 AM, Shaun Wingrin wrote:
Say, I've a SIP extension. How can I change the SIP response code to match
those needed by the registered SIP device? In this case a Mitel PBX.Tx Shaun
Why do you need to do this?
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($socket);
}
ast_claves();
?
You are closing the socket before reading the result of 'Logoff' and
Asterisk is complaining.
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?
1.6.2.19 was to be the last release of the 1.6.2 branch, so I'm not sure
if another build is expected. However the issue does reference
1.6.2.19.1 so it is possible.
However, you can see what changed between 1.6.2.18 and 1.6.2.19 in an
attempted to narrow down the bug.
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for the OS, with Linux it is not required.
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.
Even if the community does as you ask, it would not guarantee security.
Good security required upkeep and maintenance.
As an example, what version of Asterisk are you running on your
production sites?
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. Such a response SHOULD include a
Warning header field value explaining why the offer was rejected.
[1] http://www.ietf.org/rfc/rfc3261.txt
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On 11-07-23 11:48 AM, Patrick Lists wrote:
On 07/23/2011 04:00 PM, Paul Belanger wrote:
A UAS rejecting an offer contained in an INVITE SHOULD return a 488
(Not Acceptable Here) response. Such a response SHOULD include a
Warning header field value explaining why the offer was rejected
like for [599]? The problem is asterisk
(specifically app_dial, then netsock2.c) is converting 'SIP/599' to
'SIP/0.0.2.87' and sending the INVITE to that address. This is a
regression introduced when IPv6 was added.
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the problem?
I'm assuming they are still operating an Asterisk box without the
patches you have requested.
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sends, *SIP/2.0 603 Declined *to any stranger
invites because my dialplan includes Hangup(). Is there any way I can not
send a 603 declined so to mislead the probe runner?
Have you tried disabling guests?
sip.conf
[general]
allowguest=no
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twitter
On 11-07-22 09:51 PM, Alex Balashov wrote:
Paul,
Won't that just send a 403 Forbidden?
I believe so, but I was proposing a different SIP message then 603
Declined. As you mentioned, a firewall is the real solution if OP wants
to drop packets.
Asterisk is a B2BUA, not a firewall.
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Paul
tests[1] into the testsuite.
[1] http://svn.asterisk.org/svn/testsuite/asterisk/trunk/tests/fax/
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it will not be
enabled in menuselect by default.
@OP: *CLI module load app_macro.so
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in as root?
Does -devel provide it?
# yum install asterisk18-devel ?
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Paul Belanger
Digium, Inc. | Software Developer
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missing, I believe there is an issue with crowd and
something is scheduled to look at it.
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Paul Belanger
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On 11-06-08 10:34 AM, Paul Belanger wrote:
On 11-06-07 10:20 PM, Jose P. Espinal wrote:
Hello Guys,
After the Wiki was updated to the 3.5.X version, my username is no loger
available:
user: khratos
mail: j...@slackware-es.com
I had some documents on my personal space. Is there a way
On 11-06-07 02:31 AM, virendra bhati wrote:
Hi List,
Is there any way by which we can get the length of any recorded files into
seconds ?
$ sox foo.wav -e stat
[1] -
http://www.thegeekstuff.com/2009/05/sound-exchange-sox-15-examples-to-manipulate-audio-files/
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Paul Belanger
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you make clean; make
distclean before ./configure?
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Paul Belanger
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into subversion.
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Paul Belanger
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the internet link and the box due to too many sip
channels.
Do you have:
sip.conf
[general]
allowguest=no
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Paul Belanger
Digium, Inc. | Software Developer
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On 11-05-31 01:38 PM, Örn Arnarson wrote:
Does anyone have any input as to what I can try?
We use Python and StarPy extensively for the TestSuite with no buffering
issues.
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Paul Belanger
Digium, Inc. | Software Developer
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, but it didn't worked. :-(
Any other possible solution for this problem?
It is possible this is a regressions, if you roll back to 1.8.3 does it
work?
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Paul Belanger
Digium, Inc. | Software Developer
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if you have the same issue.
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Paul Belanger
Digium, Inc. | Software Developer
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Check us out at: http://digium.com http://asterisk.org
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+Information
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Paul Belanger
Digium, Inc. | Software Developer
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'Test'?
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Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com http://asterisk.org
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+Information
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Paul Belanger
Digium, Inc. | Software Developer
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the HEAD of branches/1.8.. If 1.8.5-rc2 is create,
it is because of an issue / bug was found in 1.8.5-rc1, and will include
that fix only.
If a new issue is reported after 1.8.5-rc1 and fixed in branches/1.8, it
will not be added into 1.8.5 release, but will wait until 1.8.6-rc1.
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Paul
free. But get involved! :)
[1] http://lists.digium.com/pipermail/asterisk-dev/2010-February/042387.html
[2] http://lists.digium.com/pipermail/asterisk-dev/2009-March/037262.html
[3] http://fedoraproject.org/wiki/FedoraTesting
[4] http://www.debian.org/releases/testing/
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Paul Belanger
Digium
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