On Sat, Oct 23, 2010 at 3:35 PM, Andrew Latham lath...@gmail.com wrote:
Some other people noticed that a few days ago. I think Paul was
looking at it...
Here is the thread on asterisk-dev
http://lists.digium.com/pipermail/asterisk-dev/2010-October/046697.html
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On Fri, Oct 22, 2010 at 7:10 AM, Baha @ SH i...@saudihome.com wrote:
How can I let asterisk immediately dials a trunk when off hook?
For DAHDI:
chan_dahdi.conf
[channels]
immediate=yes
channel = 1
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On Sat, Oct 23, 2010 at 2:33 AM, Baha @ SH i...@saudihome.com wrote:
I mean sip trunk
Then this functionality is dependent on the type of SIP phone you have.
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On Fri, Oct 22, 2010 at 8:53 PM, Jerry Geis ge...@pagestation.com wrote:
It seems to respawn itself. Even on a kill -9 it respawns itself.
Stop using safe_asterisk, it has logic to restart killed processes.
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you have a deadlock. Might be
worth reading doc/backtrace.txt about debugging it.
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On Thu, Oct 21, 2010 at 10:40 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
errors on line 109 - there is no 0 before $VERBOSITY as in the other lines.
More interesting is that after make samples I have no iax2 available.
What OS are you running?
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On Thu, Oct 21, 2010 at 11:05 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
What OS are you running?
If I had to guess SUSE?
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attach your config.log
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On Thu, Oct 21, 2010 at 11:15 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Adding more info :-
Ya, so that is the issue. chan_iax2 uses res_crypto, and you likely
are missing libssl-dev (openssl) on our box.
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On Thu, Oct 21, 2010 at 11:25 AM, Dave Cotton
dcot...@linuxautrement.com wrote:
Yes and ./configure and make menuselect did not signal it. :(
Did the patch at-least work for you?
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an option to disable recording the standard CDR fields?
Have you reloaded the module within asterisk?
*CLI module reload cdr
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.
However, chan_sip.so does, and it is documented.
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' is not an option for chan_iax2.so
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= Toronto:torontoisf...@192.168.1.190
[Ottawa]
type=peer
host=dynamic
username=Ottawa
secret=OttawaIsCool
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like to get this setup with ODBC.
It is very simple to get up and running. Especially if you already
have ODBC working.
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?
DNS SRV or a SIP proxy.
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, I would set
your network so only one ethernet route is active at one time, then it
is a matter or routing. If you want both ethernet ports active, then
you are doing load balancing. Something Asterisk by itself is not
strong at. Hence the SIP proxy or DNS SRV records.
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regressions. There
have been some update to RFC2833 over the last few months.
[1] http://svnview.digium.com/svn/asterisk/branches/1.6.2/main/rtp.c?view=log
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On Sat, Oct 16, 2010 at 4:59 PM, Frank Tarczynski ft...@mindspring.com wrote:
Any pointers to share?
chan_dahdi.conf
faxdetect=incoming
extensions.conf
exten = fax,1,Dial(DAHDI/4)
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, if not already enabled.
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New
a different codec (gsm) and see what happens.
Otherwise, open a support ticket with your ITSP and have them monitor
the path of the call.
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set echocancel=yes in my configs, before 1.6 it was
enabled by default.
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On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski
markm-li...@intellasoft.net wrote:
Does anyone have links to the most recent audiocodes firmware?
Why not contact Audiocodes?
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no support. Unless Audiocodec's simply wants to
charge you more money.
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On Thu, Oct 14, 2010 at 6:46 PM, asterisk asterisk aster...@ck-lee.com wrote:
Here is the sip log
487 Request Terminated, the far end is killing your session. Talk to your ITSP.
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) and debug an incoming /
outgoing call. However, if your DTMF works locally, with asterisk and
SIP phones, but does not with your provider. Then I would suspect the
issue is with your ITSP, make sure your provider is not converting
out-of-band tones to inband, or something like that.
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downloadable
PDF at http://www.asteriskdocs.org --- HTML at
http://astbook.asteriskdocs.org or see ~buybook
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a better time to call in the Retry-After header field. This
status response is returned only if the client knows that no other
end point will answer the request.
http://www.ietf.org/rfc/rfc3261.txt
Collect a SIP trace and see if a reason is supplied.
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cannot find any
reference to MySQL and the new CEL logging tool other than ODBC. Is this the
only method available to use MySQL with CEL at this time?
Looking at the CEL config files, I don't see one specifically for
MySQL. I do have it up and running via ODBC, for what it's worth.
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On Tue, Oct 12, 2010 at 10:05 AM, Stefan Schmidt s...@sil.at wrote:
so if you dont know someone in china, it would be a good idea to block
this AND set allowguest=no to prevent this in future.
And firewall your Asterisk box.
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are trying to solve a
problem, you do not need to enable debug messages.
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(which might actually, inadvertently
answer the question ;) ).
cheers,
Paul.
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this with Vicidial successfully.
cheers,
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On Mon, Sep 27, 2010 at 12:09 PM, Danny Dias ing.diasda...@gmail.com wrote:
What should i do?
Try with the lastest DAHDI version, 2.4.0.
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On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com wrote:
The same problem!
What is the output from the following?
$ ls -la /lib/modules/
$ ls -la /usr/src/linux
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$ cat /lib/modules/2.6.26-2-amd64/build/.config
r...@sangoma-testing:/home# ls -la /usr/src/linux
lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux -
$ cat /usr/src/linux/.config
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On Mon, Sep 27, 2010 at 7:36 PM, Danny Dias ing.diasda...@gmail.com wrote:
Is that Ok?
$ uname -r
Also what version of Debian?
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On Mon, Sep 27, 2010 at 9:05 PM, Jim Dickenson dicken...@cfmc.com wrote:
Do you not need to do a ./configure command before make make install? If
so issue the ./configure command again and see if that fixes the problem.
No, it does not exist for DAHDI.
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On Fri, Sep 24, 2010 at 9:11 AM, IMS ims77@gmail.com wrote:
No ideas ?
Just give me the way if possible
Download the latest asterisk version (1.4.36) and retry, if it fails
create a new issue on https://issues.asterisk.org
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On Fri, Sep 24, 2010 at 10:47 AM, Daniel Tryba dan...@tryba.nl wrote:
Am I missing something?
DEBUG_THREADS
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On Thu, Sep 23, 2010 at 10:06 AM, khalid touati khalidtou...@gmail.com wrote:
do you guys know how i can
turn debug on or just know why it's not getting enabled?
Thanks a lot for your help!
Abdullah
*CLI set debug 15
*CLI reload
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On Wed, Sep 22, 2010 at 1:27 AM, bruce bruce bruceb...@gmail.com wrote:
Any feed back is appreciated.
Then configure you endpoints to use the 192.168.100.0/24 network.
This is not an Asterisk issue, since your Aastra 55i/2.5.2.1500 is
sending the INVITE message.
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On Wed, Sep 22, 2010 at 5:42 AM, Nikhil d.nik...@cem-solutions.net wrote:
Anyone knows how to do cross compile asterisk 1.6.2.13 using
mipsel linux.?
$ ./configure --help
Will output the flags you need to set.
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On Wed, Sep 22, 2010 at 9:21 AM, IMS ims77@gmail.com wrote:
Do you have any ideas of the problem ? config.log don't give me more
explanations.
Attach your config.log so we can see what is going on.
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to https://issues.asterisk.org
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a disconnect tone? Most don't. Your best to
record the call, and analyst the tone, you can then update
indications.conf. The next best thing is to implement timeouts within
your dialplans.
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currently redirects to the proper download.
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, why are
calls going outside the VPN? Or do you have remote agents that are
not using a VPN?
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On Mon, Sep 20, 2010 at 11:48 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Anyone have a AudioCodes with Asterisk ???
Yes, but why? Both do the same thing. It would be like me asking 'I
have a bike and need to get to work. Can I use the bike with a car?'
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On Mon, Sep 20, 2010 at 7:31 AM, Arie Skliarouk sklia...@gmail.com wrote:
When call arrives from PSTN, the phone continues ringing even after caller
hanged up.
I suspect a bug [1] but without a SIP debug, I cannot be sure.
[1] https://reviewboard.asterisk.org/r/870/
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Paul Belanger | dCAP
on
http://asterisk.org. I count 5 separate marketing ads on the download
page alone. This is just my opinion.
However, on http://www.asteriskexchange.com, no problems.
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.
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On Sat, Sep 18, 2010 at 6:54 AM, unsero...@aol.com wrote:
What am I doing wrong?
Enable a SIP debug and find out.
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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On Fri, Sep 17, 2010 at 9:24 AM, khalid touati khalidtou...@gmail.com wrote:
in the dialplan, that would be a big help if you guys can help diagnose the
issue.
A debug log of the actually problem will be more helpful.
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On Thu, Sep 16, 2010 at 8:19 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I get so little output :
You are still doing it incorrectly. As said, doc/backtrace.txt has all
the required information.
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On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Is it normal that backtrace.txt is only 30K ??
Normal or not, simply post the results of backtrace.txt
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post, or reply like I did.
[1] http://www.freepbx.org/community
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/HOWTO_collect_debug_information.txt
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On Thu, Sep 16, 2010 at 12:46 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Thu, Sep 16, 2010 at 12:11 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Is it normal that backtrace.txt is only 30K ??
Normal or not, simply post the results of backtrace.txt
Please do not send me
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote:
Also, if I disable the firewall in my router I lose incoming audio and
outgoing audio works.
http://www.aocomputing.net/?p=3
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On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote:
The server is not behind NAT only the client above is
Sounds like a phone (not asterisk) issue then, make sure you have
setup your NAT and port forwarding properly on the client side.
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On Wed, Sep 15, 2010 at 9:14 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
I have no experience with this, so I post my output :
Read doc/backtrace.txt it will explain how to generate a backtrace
from a core dump.
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On Wed, Sep 15, 2010 at 11:54 AM, Ryan Wagoner rswago...@gmail.com wrote:
Anybody else notice that the 1.6.2.12 download has a version and
changelog for 1.6.2.12-rc1?
I can confirm, asterisk-dev notified.
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to define grammars into your
speech engine, it would take a large amount of work to set this up.
In the past when this has been a customer requirement, I have had to
hire a transcribing service for my audio file.
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On Tue, Sep 14, 2010 at 5:59 AM, zhou tianjun zho...@gmail.com wrote:
I want to know does the asterisk can realize
that. Or I
have to write module for that function ?
No, you need to tell Asterisk what to do.
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On Tue, Sep 14, 2010 at 2:27 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
And again !! Without me doing anything !!
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
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to this thread, also how are you setting up DTMF,
inband or rfc2833?
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On Mon, Sep 13, 2010 at 11:22 AM, Bryant Zimmerman brya...@zktech.com wrote:
Is there a way to drop a ip connection to asterisk after a number of
register attempts.
Not within Asterisk. Google fail2ban
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On Mon, Sep 13, 2010 at 2:53 PM, Jonas Kellens jonas.kell...@telenet.be wrote:
anyone on this list knows how to turn these messages off please ?!
*CLI core set verbose 0
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On Tue, Sep 14, 2010 at 1:01 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
Is it possible to record say 30 seconds of audio and then have LumenVox
convert to text ?
ASR, yes.
http://www.digium.com/en/products/software/lumenvox.php
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a new issue on
https://issues.asterisk.org
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.
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On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com wrote:
Sending to 123.123.123.123 : 5060 (no NAT)
Either you changed the peer parameters or they did...
If he is not receiving any response, it is most likely a routing issue.
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the actually issue, if the OP does the following:
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt
The attached the debug log to thread.
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.
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On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
asterisk*CLI core show application Dial
did you have libxml-doc installed when you build asterisk?
*CLI module load app_dial.so
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On Thu, Sep 9, 2010 at 11:37 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
did you have libxml-doc installed when you build asterisk?
s/-doc/-dev
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On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Can I install libxml-doc now without having to rebuild asterisk ?!
No, install libxml-dev then rerun ./configure, make install
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Progress SIP Message.
Usually used to play audio to the line, by passing toll charges as a
result of using Answer().
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On Thu, Sep 9, 2010 at 4:40 PM, Tim Nelson tnel...@rockbochs.com wrote:
Can anyone tell me how to build fxstest?
No, but if you output the error message we can help point you in the
right direction.
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Paul Belanger | dCAP
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New
)
exten = 22,n,Read(roomid,conf-getconfno,6,1)
exten = 22,n,MeetMe(${roomid},Ms)
exten = 22,n,Hangup()
exten = i,1,Playback(conf-invalid)
exten = i,n,Goto(22,1)
exten = t,1,Goto(22,1)
Your dialplan is missing priory 1 for you Answer().
exten = 22,1,Answer()
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Paul Belanger | dCAP
Polybeacon
On Tue, Sep 7, 2010 at 2:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Appreciate your kindly advise and help.
With the proper dial-plans, testing and additional development,
everything you have listed is possible.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan
within features.conf
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Paul Belanger | dCAP
Polybeacon | Consultant
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On Thu, Sep 2, 2010 at 2:26 PM, Thorolf Godawa nos...@godawa.de wrote:
Any idea what is going wrong here?
Read doc/backtrace.txt
If you cannot get Asterisk to coredump, try running it under gdb to
see what is happening.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan
On Thu, Sep 2, 2010 at 8:31 AM, Paddy Grice pa...@wizaner.com wrote:
Any suggestions
You have to modify the source code.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
On Tue, Aug 31, 2010 at 4:04 AM, Alex Ferrara a...@receptiveit.com.au wrote:
exten = 849,1,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
exten = 849,1,Progress()
exten = 849,n,Playback(custom/ceh-meetingmsg)
exten = 849,n,Hangup
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber
On Tue, Aug 31, 2010 at 5:57 AM, jordan pan pylon...@gmail.com wrote:
my asterisk will coredump in runing about ten days one time, and the
following is bt infor:
Open an issue on https://issues.asterisk.org, besure to follow
doc/backtrace.txt and post all relevant information.
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Paul
by entering:
*CLI dialplan show 6789542...@remote
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New to Asterisk? Join us
On Tue, Aug 31, 2010 at 5:50 PM, Alex Ferrara a...@receptiveit.com.au wrote:
Hi Paul,
I tried adding Progress() to no avail. I still get no audio and below is what
comes up in the console.
Try moving Progress() before the Dial(). If you Answer() the channel,
do you have the same problem
Specifically,
*CLI sip set debug peer xxx
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
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/HOWTO_collect_debug_information.txt
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Paul Belanger | dCAP
Polybeacon | Consultant
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blog.polybeacon.com
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New
extension not found in context
'extensions.conf'.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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On Mon, Aug 30, 2010 at 2:48 PM, Todd Reese trees...@gmail.com wrote:
Thanks for pointing out the misspelling. I've corrected that and still no
luck.
Create a new debug log with your recent changes, re-attach it the list.
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan
On Sun, Aug 29, 2010 at 10:52 PM, Zhang Shukun bit...@gmail.com wrote:
but i want to know if i can invite some one to the conference when i
already in the conference?
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber
application Dial
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
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New
On Fri, Aug 27, 2010 at 12:46 PM, jeremy.hellst...@synovate.com wrote:
moving the dchannel around, 12 through 24. Does anyone see anything
blatantly wrong?
What alarms are you getting?
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Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger
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