a branch I created a few months back, and I have not looked
at it in a while. That said, there maybe issues with it.
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' status is 'UNKNOWN'
Asterisk 1.8? Are you using realtime? Looks to be an issue with
netsock2.c.
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On 11-04-08 11:55 AM, satish patel wrote:
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
Best you can do is collect a full debug[1] log and see when the issue
On 11-04-08 12:56 PM, Paul Belanger wrote:
On 11-04-08 11:55 AM, satish patel wrote:
@Paul - many time i am gettting following SIP error when channel isn't
available. I want to get rid on this revers thing. I tried all version
1.8.1,1.8.2,1.8.3 but not fix :(
Best you can do is collect
method at the moment is to look at the code,
not the best solution.
I'd like to add them for Asterisk 1.10, on my TODO list. Maybe I'll do
some work on them this weekend; depends on the weather :)
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you done with Asterisk? Blindly moving into production with _anything_
is a recipe for trouble.
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russellb's branch [1].
[1] http://svn.digium.com/svn/asterisk/team/russell/messaging/
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. :)
[1] http://svn.digium.com/svn/testsuite/asterisk/trunk/
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Just a guess but is it possible one of your SIP peers (7623 or 7624)
has an invalid IP address of 0.0.29.200? I wonder what sip show
peers shows.
On Thu, Apr 7, 2011 at 4:03 PM, satish patel satish...@hotmail.com wrote:
Re-opening this issue.
If i dial number which doesn't existing then i
that, I've
been testing it with 1.8 for a while; under light load (less then 5
channels). I have not had a problem.
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First, this appears to be working for me though I'm not 100% sure of
that and cannot guarantee it will for you in any way, shape or form.
With the lawyering out of the way...
I've seen fail2ban allow more than 500 failed SIP login attempts in
under 30 seconds before adding an iptables rule to
? How do I get this to be consistent?
Have you included the context properly?
[mndemo_to_mediaport105]
include = smvoice-mediaport
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it to subkeys.pgp.net yet.
[1] - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
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On 25/03/11 14:36, Douglas Mortensen wrote:
Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like
to ask the question as to why I should NOT use 1.8? What are specific reasons? For
instance a few days back I was speaking with James at Rhino Equipment. He said
On 24/03/11 05:49, Olivier CALVANO wrote:
The To, To:sip:081169x...@91.121.xxx.xxx;user=phone, can i get it into
a variable for sent it at a API ?
You want the sip_header function:
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
cheers,
Paul
not.
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On 11-03-25 03:13 PM, satish patel wrote:
Both servers files are identical..
*CLI module show like cli
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://issues.asterisk.org/view.php?id=18818
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straight out again over a
SIP or IAX trunk.
Also, ALL the contexts that are being used are realtime - they are not a
mixture.
Do you think it could still be related?
We'd need to see the backtrace, but possible.
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of sip-vicious, you
can often stop such things by using the svcrash.py script they now
provide.
Check your iptables logs.
cheers,
Paul.
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.
Collect a complete debug log[1] and open a new issue on the tracker.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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is correct, your issue does look to be a dialplan problem
(SIP/h). I should have been more specific about handling the failure
more gracefully.
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/Zaptel close the channel after the remote end has hung up?
Is this an analog line? If so, is your CO providing a disconnect tone?
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to
understand how pattern matching works with Asterisk will help you in the
long run.
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On 11-03-15 05:52 PM, deeps backup wrote:
Does your migration plan include testing asterisk 1.8 before moving it
into production?
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in your dialplans to share
information before the Asterisk boxes.
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/testsuite/asterisk/trunk/
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it upstream
(issue tracker), and we cab merge it into the testsuite. Then every
each time svn commit is made, http://bamboo.asterisk.org will run the
testsuite against it.
It is beneficial to both you and the community :)
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On 11-03-09 12:16 PM, Jerry Geis wrote:
Does asterisk 1.8 still need dahdi installed if you only doing SIP and
ALSA/console.
Only if you plan to use MeetMe().
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] https://issues.asterisk.org
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On 11-03-03 03:20 PM, bayardo.sanc...@gmail.com wrote:
I listened to your email using DriveCarefully and will respond as soon as I
can.
Download DriveCarefully for free at www.drivecarefully.com
Please disable your auto responder.
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twitter
On 11-02-23 05:39 AM, Ishfaq Malik wrote:
Has anyone else experienced anything like this?
There is a patch on the issue tracker[1], please test it out and report
your feedback.
[1] https://issues.asterisk.org/view.php?id=18168
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.
I've tried adding name, username, auth, authname, defaultname, and
fromname among others and none seem to do what I'm looking for. Tried
reading through chan_sip.c but have not figured it out yet. Figured
I'd ask.
Thanks in advance,
Paul
PS: I had a SIP entry with too easy a secret get exploited
realtime? It would be good to see a
full debug[1] log of your call.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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On 11-02-17 12:06 PM, viswavardhanreddy karna wrote:
Hi all,
I have a few doubts regarding asterisk...
You are correct, Asterisk is a B2BUA, not a SIP Proxy.
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:)
Is there such application in standard asterisk's applicartions ? And if
no maybe there is some add-on.
*CLI core show function PITCH_SHIFT
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Jared Smith.
[1] http://lists.digium.com/pipermail/asterisk-users/2009-July/234419.html
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] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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On 11-02-11 04:37 AM, Ishfaq Malik wrote:
Does anyone have any rough idea how far away 1.8.3 is?
If you are hard up for a release, you can use the latest 1.8.3 RC[1].
[1]
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar.gz
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, if you are waiting for asterisk-1.8.x to be released, start
testing asterisk-1.8.x-rcx and report feedback.
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= (24 channels)
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On 11-02-11 04:15 PM, satish patel wrote:
That means 1 card 1 span 2 card 2 span
Not really, for example 1 card can have 4 spans (Digium TDM410P). You
can usually substitute the word 'port' for 'span'. EG: My card has 2 T1
ports - My card has 2 T1 spans.
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of
run-of-the-mill Linux, and I would like to avoid having to patch
Asterisk.
Should be no more then 10-15 lines within your extensions.conf
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On 11-02-01 04:02 AM, Felix Dong wrote:
I got a question to asterisk 1.6. Is it possible to playback a Audiofile in
uplink and to record the downlink channel in another Audifile at the same
time?
Yes, look at MixMonitor.
*CLI core show application MixMonitor
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to use this approach for the test. I was able to
find some limited information on the wiki, let me see if I can get it
working.
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This is the correct method. But you are saying even if you stop and
start Asterisk res_musiconhold.so is still loads?
I just tested with the latest 1.6.2 branch with the same settings, MOH
was not loaded.
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On 11-01-27 11:41 AM, Amardeep Rana wrote:
Please give idea for Multi tenant with Trixbox or elastix.
http://astbook.asteriskdocs.org
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would not
work on anything else.
[CREATECALL]
dsn=Example
writesql=INSERT INTO x (y) VALUES (z)
readsql=SELECT LAST_INSERT_ID();
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LAST_INSERT_ID();
Hey, thanks for the tip Paul!
I should also note, make sure you create a 2nd DSN for your specific
ODBC commands that will use LAST_INSERT_ID(), otherwise if you are using
ODBC CDR or CEL, there is a chance LAST_INSERT_ID() will return the ID
of those records.
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Paul
that polls the
AMI for information, then sends it to the load balancer.
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.
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handle_request_invite: Call from '' to
extension 'service' rejected because extension not found in context
'default'
You can troubleshoot dialplan issues pretty easy with:
*CLI dialplan show service@default
The message states, exten = service,1,blah() is missing from the
[default] context.
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On 11-01-23 10:01 PM, RR wrote:
I'm sure this has been talked about and based on some searching of archives,
I'd discovered that to be able to use timerfd, one needs to have a kernel
version =2.6.27? Is this true?
Kernel version 2.6.25 or newer, as documented in CHANGES.
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-1
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(${ISNULL(${SCHEDULEID})})
func_odbc.conf
---
[CHECKCALLBACK]
dsn=asterisk
readsql=SELECT s.Id, c.CallDate, c.Id FROM Schedules s, ListContacts c
WHERE s.user='${SQL_ESC(${ARG1})}' AND s.Contact=c.Id AND s.Active=True
Hope it helps.
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as a result of viruses.
It is recommended that you should carry out your own virus checks
before opening any attachments.
Registered in England. No. 27459085.
Additionally, do you really need a 17 line[1] signature?
[1] - http://s3.amazonaws.com/theoatmeal-img/comics/email/4.png
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them.
[1] http://www.asterisk.org/community/rules
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can run:
*CLI dialplan show 2103@incoming context
to see if the dialplan actual exists.
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On 11-01-16 03:58 PM, Thomas Perron wrote:
Does anyone see any issues here? I cannot get it to work.
Passwords are not real!
No, however you did not provide any debug logs [1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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://downloads.digium.com/pub/telephony/fax/README
[2] - https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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, with no tif
written where I've directed it to.
Looks like a problem related to ast_verbose(), without looking at the
source code for res_fax_digium.c, a workaround maybe to disable verbose
output. Not a pretty solution, but may help the crashes.
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://www.asterisk.org/community/rules
[2] http://linux.sgms-centre.com/misc/netiquette.php#toppost
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+Supplementary+Services+%28CCSS%29
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On 11-01-07 01:33 PM, Bruce B wrote:
Where can I find this?
manager.conf.sample?
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New
to see a full debug[1] log.
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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. Please open a new issue on the tracker, include your debug log
and sip.conf.
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.
Also, once you located your core file, generate a backtrace[1] and
attached it to this message.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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] :) I've already committed[2]
the code to all branches. The issue is will chan_iax2 destroying the
scheduler too soon.
[1] http://bamboo.asterisk.org/
[2] http://svnview.digium.com/svn/asterisk?view=revisionrevision=296670
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to go from here...this is the excerpt
from the netsock.c file:
I'm in the process of bring up our remote Bamboo agents for Solaris, so
I can see if I get the same issue. Which versions of Solaris and
OpenSolaris are you using?
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On 10-12-02 12:22 PM, Bryant Zimmerman wrote:
Karsten
I do not see it in the changlog for the 1.8.1 rc version.
How would I get the SVN version to test?
$ svn co http://svn.asterisk.org/svn/asterisk/branches/1.8
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to dialplan variables.
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On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote:
make[1]: *** [cdr_webservice.o] Error 1
make: *** [cdr] Error 2
What is cdr_webservice.o ?
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of the documentation process, I would
recommend talking to some on #asterisk-dev.
It is my understanding there will be workflows around the wiki content
and a reviews process for new submissions.
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On 10-11-21 09:41 PM, Jonathan C. Bailey wrote:
Does anyone know what changed between 1.4 and 1.8 in regards to early audio
(both hearing it and interacting with it)?
Read UPGRADE.txt and CHANGES
*CLI core show application Progress
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/1.8
$ cd 1.8
$ ./contrib/scripts/live_ast configure
$ make
$ ./contrib/scripts/live_ast install
$ ./contrib/scripts/live_ast samples
$ ./live/asterisk -vc
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backend (MySQL). A static configuration of users in sip.conf resolves this
and everything works fine.
What version of 1.6? I _think_ this may have been a bug, that was fixed.
Don't hold me to that.
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On Wed, Nov 10, 2010 at 2:11 PM, Jose P. Espinal j...@slackware-es.com wrote:
Is it possible to select ODBC_STORAGE without entering to 'menuselect'?
$ ./configure
$ make menuselect.makeopts
$ menuselect/menuselect --enable ODBC_STORAGE menuselect.makeopts
$ make
...
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On Wed, Nov 10, 2010 at 2:42 AM, Olle E. Johansson o...@edvina.net wrote:
Any update, Paul?
Yes, still reproducible. I'm going to do some homework first before
creating a new issue on the tracker. Specifically, if anybody else
can reproduce it.
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/~pabelanger/Compiling+Asterisk+for+Developers
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.
Only compile the modules you need.
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. Most desktop
distros are just too bloated for an embedded solution.
[1] http://www.askozia.com/
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On Fri, Nov 5, 2010 at 4:44 PM, bakko asannu...@gmail.com wrote:
When I load res_ais.so module, the pbx crash (boths)
Generate a backtrace[1] and upload to this thread.
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
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Jabber
but not 1.8, please
created a new issue on the tracker and we will triage it. Also
include a debug log [1].
[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
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On Wed, Nov 3, 2010 at 9:18 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
exten = s,n,Set(vgLabel=vg(${number}+1))
exten = s,n,Set(vgLabel=vg$[${number} + 1])
untested
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Paul Belanger | dCAP
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On Tue, Nov 2, 2010 at 8:29 PM, Dan Journo d...@keshercommunications.com
wrote:
Or does this kind of thing need a serious network switch?
Why not set MLPPP, assuming your provider supports it.
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Paul Belanger | dCAP
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On Tue, Nov 2, 2010 at 3:20 PM, Silver Thorne zora...@gmail.com wrote:
Any help at all would be appreciated.
DUNDi
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Paul Belanger | dCAP
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(google apps) is how. I haven't had much time to
debug it, but plan to play more with it the coming weeks.
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Paul Belanger | dCAP
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users are not going to understand...
Had the same issue, but have not had a chance to find a good solution.
You could change your status to DND. I tried invisible put seems not
to be supported.
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Paul Belanger | dCAP
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).
I managed to finally get a GV number while at Astricon. I hope to
play with this more next week.
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Paul Belanger | dCAP
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On Tue, Oct 26, 2010 at 8:26 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
I'm going to try and look at this during Astricon :)
Ok, just uploaded a new patch on
https://issues.asterisk.org/view.php?id=18202 Let me know if it
worked.
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Paul Belanger | dCAP
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(): ai_family not
supported
I'm going to try and look at this during Astricon :)
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Paul Belanger | dCAP
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://svn.asterisk.org/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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Paul Belanger | dCAP
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Paul Belanger | dCAP
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with realtime IAX2 yesterday, do you
mind trying the branch listed on the issue tracker.
Also, please post a copy of your iax2.conf file that will reproduce
the issue (mask any passwords).
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Paul Belanger | dCAP
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. :)
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Paul Belanger | dCAP
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On Sat, Oct 23, 2010 at 3:03 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
Okay, just reproduced your issue and looking at the code now. :)
Ok, think I fixed it. You can either apply this patch to 1.8.0, or
svn update the branch I'm working on. Feedback is welcome.
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Paul Belanger
1.6.2 for the practical. And I believe
there were 1.6 questions on the written too.
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Paul Belanger | dCAP
Polybeacon | Consultant
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