On 5/21/18 1:49 PM, D'Arcy Cain wrote:
I am having troubles with sending faxes. I hope someone can help me
work out a better method.
I have a project that I like to use to send faxes. It might be able to
drop into your environment pretty easily.
https://github.com/jkister/astelegraph
I
isk 14. the project
certainly needs more hands on the code.
regardless of asterisk14, an important bugfix is at
https://github.com/jkister/app_swift but darren has not
accepted/rejected the changes.
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om) and a few contexts confuzzled
(missing general/globals and extra parkedcalls - but again I get it) -
it seems to be perfect.
One for a wiki, somewhere.
thanks,
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On 4/13/16 11:57 AM, A J Stiles wrote:
You could try
*CLI> dialplan show
Between my older backup and dialplan show, I guess that's my best shot.
Thanks :D
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On 4/13/16 11:37 AM, Steve Edwards wrote:
Will 'dialplan save' help?
I just tried this one. It writes the dialplan, but without the
application arguements. Worthless.
right, was a good shot. in my case I have writeprotect=yes in general,
so that would have been the first hurdle. but
with the slip of a finger, i destroyed by extensions.conf (grep -i >
extensions.conf)
I have a backup that is dozens of hours of code old.
is there a way i can use the asterisk cli (or some other asterisky
method) to recreate that extensions.conf ?
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let me know if you find it useful,
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e like I could with Page.
Is there an easier replacement of app_page ? I'd hate to keep
dahdi+meetme just for Page.
I would post here what I have so far, but it's so complex it would be a
headache to explain what I was thinking.
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p
__EOE__
cat <<__EOS__ > /var/spool/asterisk/tmp/test123
Channel: Local/221@intercom
Callerid: "TTS" <0>
MaxRetries: 2
WaitTime: 45
Context: tts
Extension: s
Priority: 1
__EOS__
mv /var/spool/asterisk/tmp/test1
Using Asterisk 13.3.2 on CentOS7 and pjproject 2.4, I can't make
asterisk try to send a register.
I have configured my pjsip.conf similar to
For anyone interested, Allison Smith's AMA (not sure she's still around):
http://www.reddit.com/r/IAmA/comments/2rrb7m/iama_professional_telephone_voice_ama/
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On 10/14/2014 2:25 AM, chandapure shiva wrote:
I have put nat =force_rport,comedia in general section , but still not
working .
I hate to ask, but did you reload sip afterwards? asterisk -rx 'sip reload'
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in sip.conf ?
since you have the 'stun show status' command, i beleive the correct nat
statement is nat=force_rport,comedia in the general section.
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and report back any issues.
git clone 'https://github.com/darrensessions/app_swift'
cd app_swift
configure
make
make install
make reload
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named uniquely ?
there are bugs (e.g., jira# 11291) that have to do with files having the
same name.
my solution was to add .$$ on the end of the filename to ensure it was
unique.
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is that my MUA doesn't support bottom-posting, which holds no water.
i dont care that much, though- i don't waste time on top-posted messages
a nor messages that are quoted stupidly.
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and was
working on forking it as an official version.
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I have regularly (once a week, once per few hundred calls?) been having
problems with Asterisk's SIP stack not responding to packets from any of
my registered devices. In the past, I could not tolerate the outage, so
i would restart asterisk to make things happy.
My Asterisk server is
On 11/12/2013 7:37 PM, Jeremy Kister wrote:
any ideas how we can find out what's upset ?
more info:
when I create a /var/spool/asterisk/outgoing/callfile (with multiple
SIP/xxxSIP/yyy), the extensions ring. but when i answer with the
handset the call does not connect and the other
On 11/12/2013 8:46 PM, Duncan Turnbull wrote:
Any chance DNS is dying about the same time the problem occurs
good idea, but I don't use DNS anywhere in Asterisk. well, except for
sip.conf:externhost. it's all IP addresses.
--
debug peer vgw1' on the console. but i dont see what's
causing the issue..
http://kister.net/tmp/ast-sip.conf
http://kister.net/tmp/ast-console.txt
can anyone spot the issue?
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just thought this was cute enough to pass along,
https://www.youtube.com/watch?feature=player_detailpagev=GLwct15X_3g#t=135
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section and/or each peer in sip.conf:
session-timers=refuse
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asterisk is started, perhaps try core set verbose 10, core set
debug 10, module unload chan_sip.so, and module load chan_sip.so . if
there are any errors loading the module it may be easy to spot them.
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On 5/9/2013 3:13 PM, asterisk...@jeremykister.com wrote:
I frequently see on the console:
WARNING[7832]: chan_sip.c:19134 show_chanstats_cb: Could not get RTP stats
bump. (sorry).
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On 5/9/13 8:21 PM, Brian LaVallee wrote:
When qualify is enabled on a trunk, the From line shows asterisk. See the
SIP message below.
I had the same annoyance/issue. fixed it in
https://issues.asterisk.org/jira/browse/ASTERISK-17616
the patch was included in 1.8.9 rc1.
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with call quality whatsoever.
i'm running sip image 03-08-12
g711ulaw only.
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4573 and saw that Asterisk is
not attempting the connection.
can someone replicate this behavior ? Or is this just my config ?
* http://jeremy.kister.net/code/asterisk/simple_agid.pl
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On 2/11/2013 11:13 PM, Jeremy Kister wrote:
[Feb 11 16:28:45] ERROR[28501][C-0012]: utils.c:1187
ast_carefulwrite: write() returned error: Connection refused
[...]
can someone replicate this behavior ? Or is this just my config ?
opening issue in jira; this is a bug.
https
On 1/26/2013 4:00 PM, Richard Mudgett wrote:
features. You have found two bugs in confbridge:
Issues created in jira. thanks for your input!
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conf_invalid_event_fn: Invalid event for confbridge user ''
[Jan 25 23:50:52] ERROR[3745][C-000a]: confbridge/conf_state.c:47
conf_invalid_event_fn: Invalid event for confbridge user ''
any way to hush/fix that?
Thanks,
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, regardless of misspellings.
I could be convinced to vote up 1s for I, 0s for O, and 3 for E. So
SIP_CODEC, S1P_C0D3C, and SiP_cOdEC would all evaluate equally. The
next step would be to appease the English spelling reform people by
allowing SIP_KODEK too. :p
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be useful out of the box, depending on what you're
trying to do.
http://jeremy.kister.net/code/asterisk/jkSMS
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have replication configs at http://jeremy.kister.net/tmp/ast/
Can someone help me determine if this is a problem with asterisk or ios ?
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)
But it didnt help, still randomish stutter lining up with the disk.
this is a great help, at least i can start hacking at things now.
Thanks,
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% cpu.
Is this behavior indicative of a timing problem? loading
res_timing_pthread.so makes things horribly worse. i don't believe any
other software timer is available for Solaris/sparc, right ?
other thoughts ?
Thanks,
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On 7/18/2012 2:27 AM, Jeremy Kister wrote:
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
.. ok, if the system weren't Solaris - let's say it was Debian Linux,
what would be on the list of things to check for ?
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/features.conf ? perhaps to put
the caller on hold ?
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On 6/22/2012 10:39 PM, Darren Sessions wrote:
both would be appreciated.
if you can send me a backtrace, that'd be great
http://jeremy.kister.net/tmp/swift/
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i just compiled app_swift 3 from the new git repo for asterisk 1.8.13.0
asterisk loads the module fine, but as soon as i try to swift anything,
asterisk core dumps.
i'll be glad to post the corefile or sample extensions.conf if desired.
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on
iPhone. They also ahve software for Android, but I cant attest.
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=dynamic
deny=0.0.0.0/0
permit=192.168.0.0/24
then asterisk -rx sip reload
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?
if it's an older version of 1.8 ( 1.8.4) and you're also recording the
call, you may be encountering a known bug.
https://issues.asterisk.org/jira/browse/ASTERISK-17346
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for the heads-up.
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(confbridge:${CONFNO})}?1)
exten = s,n,Set(DB(confbridge/${CONFNO})=1)
[foo]
exten = s,1,Macro(confbridge-setup)
exten = s,n,ConfBridge(${CONFNO})
exten = s,n,NoOp( ${DB_DELETE(confbridge/${CONFNO})} )
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On 12/19/2011 4:08 PM, Asterisk Development Team wrote:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.22
or for the non-404-version:
http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ChangeLog-1.6.2.22
;p
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On 12/9/2011 12:55 AM, Mike Diehl wrote:
What am I doing wrong?
perhaps try: http://jeremy.kister.net/code/asterisk/mwi.pl
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are working fine.
anyone have suggestions on what i can try next?
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to allow reinvite on s3? or
is this something that should go to the tracker ?
thanks,
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On 10/10/2011 10:08 PM, Andres wrote:
I would recommend Acrobits. Not free but only a few bucks. It works
fine with ATT 3G.
+1
only thing i like better is it's big brother, Groundwire
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parts of your dialplan and sip.conf ?
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, AUTHORITY: 0, ADDITIONAL: 0
;; QUESTION SECTION:
;ilbcfreeware.org. IN NS
;; Query time: 15 msec
;; SERVER: 205.178.190.42#53(205.178.190.42)
;; WHEN: Fri Sep 2 16:03:41 2011
;; MSG SIZE rcvd: 3
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On 9/2/2011 4:15 PM, Jeremy Kister wrote:
since www.ilbcfreeware.org is broken, asterisk installs that want ilbc
are failing.
it appears this was done on purpose since Google bought them.
Asterisk is going to need fixing. I'll probably hook something up.
http://www.webrtc.org/ilbc-freeware
On 9/2/2011 8:33 PM, Jeremy Kister wrote:
Asterisk is going to need fixing. I'll probably hook something up.
https://issues.asterisk.org/jira/browse/ASTERISK-18412
a patch and brief instructions are now available at the above URL.
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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
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.
http://www.cyberciti.biz/faq/linux-port-redirection-with-iptables/
remember UDP vs TCP.
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to the
$CHAN variable.
adding | tr -d '\n' to the end of the command fixed it right up.
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anyone from digium around ?
https://issues.asterisk.org/jira/
Oops - an error has occurred
System Error
Cause:
java.lang.NoClassDefFoundError: Could not initialize class
org.codehaus.xfire.util.STAXUtils
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as me
being the reporter in jira? or are the tickets going to stay in mantis
until they are resolved and never make it into jira ?
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On 6/6/2011 1:08 AM, Jeremy Kister wrote:
similarly, are tickets that I reported in mantis going to show as me
being the reporter in jira? or are the tickets going to stay in mantis
until they are resolved and never make it into jira ?
after some more clicking, i see the answer to this one
/code/asterisk/iptables.init
modify RTPRANGE and the trusterd array at the top,
add in your DID providers to the siprtp array at the top,
that should get you near there.
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On 5/14/2011 9:45 PM, Jeremy Kister wrote:
http://jeremy.kister.net/code/asterisk/iptables.init
oops, that's:
http://jeremy.kister.net/code/iptables/iptables.init
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] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to establish the call anyway
the calling side just hears ringing.
i have plenty of debug info, but nothing too interesting. anyone else
having this problem ? or is it time for bug report ?
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On 5/12/2011 11:08 PM, Jeremy Kister wrote:
[May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to establish the call anyway
I found the problem, and I am sending in a bug report :)
if anyone is interested, the issue is 19286 (i'll
://issues.asterisk.org/view.php?id=18742
didnt make it into this 1.8.4 release ?
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ConfBridge with app_page ?
then i could disable meetme dahdi_dummy all together.
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tried with verbose 10/debug 10 before posting. no dice.
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]
[logfiles]
console = notice,warning,error,dtmf
messages = notice,warning,error,verbose,dtmf,fax
if i send 'options' or 'register' from a non-configured sip peer, i dont
see anything in the log. am I missing something ?
* i can replicate this behavior on 1.8.2.3 and 1.8.3.2
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On 4/16/2011 8:20 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
stop it.
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see anything in the log. am I missing something ?
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On 4/15/2011 3:39 AM, Jeremy Kister wrote:
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
forgot to mention that I can replicate this behavior on 1.8.2.3 and 1.8.3.2
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in the if [ $THROTTLE ] section.
if not, just:
# make-non-na.pl
# vi iptables
## change the MYLAN=10.0.0.0 to whatever you use
## change the RTPRANGE to whatever you have in rtp.conf
# mv iptables.init /etc/init.d/iptables
# /etc/init.d/iptables start
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OPTIONS
Server: DC-SIP/2.0
Organization:
Content-Length: 0
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) to contain the user you want
uhm, didn't I ?
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.
https://issues.asterisk.org/view.php?id=19036
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=${CHANNEL:4})
exten = s,n,Set(XTN=${CUT(STRIPPED,-,1)})
exten = s,n,Macro(Jabber,x${XTN} answered ${CALLERID(num)})
[macro-Jabber]
; ${ARG1} - message
exten = s,1,Jabbersend(m...@example.com,m...@example.org,${ARG1})
exten = s,n,Jabbersend(m...@example.com,m...@example.net,${ARG1})
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http
On 3/15/2011 11:18 AM, Paul Belanger wrote:
Theses are leftover issue with the IPv6 conversion for Asterisk 1.8.
Collect a complete debug log[1] and open a new issue on the tracker.
I believe one was entered a few months ago-
https://issues.asterisk.org/view.php?id=18514
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any issues.
it's not for the faint-of-heart and might require a bit of hacking
(really minimal though) if you're not running the same tools that i'm
running (like editing the code's DSN if you dont have sqlite installed)
http://jeremy.kister.net/code/asterisk/jkSMS/
enjoy,
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})
exten = s,n(snooping),NoOp(snooping on ${CHANNEL})
that'll end up putting a mp3 of the call in
/var/spool/asterisk/monitor//MM/DD/HHMMSS-CALLERID.mp3
don't forget any legal issues you might have to work around, recording
the fact that you declared the message is being recorded.
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On 3/2/2011 9:46 AM, Leif Neland wrote:
Some of the phones are being disconnected with Asterisk saying no reply
to critical packet
What kind of phones are they? I might have nothing to do with your
network configuration; try adding to sip.conf [general]:
session-timers=refuse
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', the call is dropped instantly.
Am I using SoftHangup incorrectly?
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On 1/26/2011 3:18 PM, Asterisk Development Team wrote:
* Reimplemented fax session reservation to reverse the ABI breakage
introduced
in r297486.
(Reported by Jeremy Kister on the asterisk-users mailing list. Patched by
mnicholson)
I can confirm that this resolves the issue
Since digium is apparently blind to users of their Free Fax for
Asterisk, does anyone have advice on how to report a crashing problem
with res_fax_digium and Asterisk 1.8.2 ?
I have detailed logs/reports and a backtrace ready, but I have no idea
who can help.
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ReceiveFax is called, asterisk crashes, with no tif
written where I've directed it to.
I have several files including backtraces and config files at
http://jeremy.kister.net/tmp/fax/
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://issues.asterisk.org/view.php?id=18194
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temporarily at http://jeremy.kister.net/tmp/modules.conf
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the same issue.
Just add it to the list of things to fix in 1.8..
Do you want to add it to http://issues.asterisk.org ?
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hosts based on X authentication attempts (good OR bad)
(fail2ban only counts bad attempts)
* this cannot detect encrypted attempts (SIPS), fail2ban can
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important. if not, the iptables.init script can likely drop in place.
if you only need north-american ip addresses to talk to your asterisk
box, i suggest you also run the make-non-na.pl from cron every week.
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deal with this?
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the packaged app_fax_spandsp and also Digium's
app_fax_digum for 1.8.0-rc1 -- no difference in behavior.
Anyone have any ideas how I can get this fixed?
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Jeremy Kister
http://jeremy.kister.net./
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On 11/13/2010 4:36 AM, Jeremy Kister wrote:
When a caller connects, asterisk switches to the fax context and hangs
up the call.
I was wrong, asterisk does not even switch to the fax extension-
i added a noop, and it's not making it:
exten = fax,1,NoOp( in fax extension )
exten = fax,n,Goto
/lib/asterisk/mohmp3
sort=random
Anyone have ideas if/how I can change this behavior?
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Jeremy Kister
http://jeremy.kister.net./
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New to Asterisk
On 11/4/2010 5:07 PM, Warren Selby wrote:
It is because you're using quietmp3 as your mode.
Can you explain what the processes are doing?
killing them doesn't affect music on hold or any other mp3 playback.
strace shows that their behavior doesnt change during a call.
--
Jeremy Kister
does do something, and the
parent is in a continuous loop making sure the child is alive.
So for the archives, i suppose asterisk spawns these off once so that it
can be used as a single source for all channels, rather than spawning
off one mpg123 for each channel.
--
Jeremy Kister
http
just put 184.72.221.84 in the
siprtp section of the iptables script.
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Jeremy Kister
http://jeremy.kister.net./
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New to Asterisk? Join us for a live
On 9/4/2010 1:31 AM, Jeremy Kister thought:
On 8/29/2010 3:25 AM, Jeremy Kister wrote:
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a critical packet being missed.
hmm, either no one
?
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Jeremy Kister
http://jeremy.kister.net./
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the
above swift --voices command ?
also, if you're going to be dialing digits with swift, you'll probably
run into detection issues unless you use my patch at
http://jeremy.kister.net/code/app_swift-1.6.2.patch
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Jeremy Kister
http://jeremy.kister.net
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