This is what comes with voicemail.conf.sample - works for me!
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
; Additionally, on forwarded messages, you have the variables:
; ORIG_VM_CALLERID, OR
Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/
On 16 February 2017 at 13:04, Max Grobecker
wrote:
> Hello,
>
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in
Hi there;
2 linux boxes and Windows all report an error and the archive is not
extractable.
Wget reports the size as follows:
2017-02-14 08:36:21 (7.29 MB/s) - ‘asterisk-14-current.tar.gz’ saved
[40653605/40653605]
It starts un-tarring but then
asterisk-14.3.0/bridges/bridge_native_rtp.c
a
Did you actually do "make install" after doing "make"?
On 7 December 2016 at 12:17, Tzafrir Cohen wrote:
> On Wed, Dec 07, 2016 at 09:23:30AM +, k...@mayten.sch.bme.hu wrote:
>> On 2016-12-07 09:13, Steve Howes wrote:
>> >On 07/12/16 04:56, christopher kamutumwa wrote:
>> >>Ive installed aste
I think it might be related to this?
https://issues.asterisk.org/jira/browse/ASTERISK-26391
I think I remember having to edit logger.conf - this is what mine
looks like now:
console => notice,warning,error
messages => notice,warning,error
Try that, restart asterisk and see if it works :)
On 30 N
Thanks for the super-quick answer! Now I was able to find this:
https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk#BuildingandInstallingAsterisk-Buildingfornon-nativearchitectures
I had just assumed a cloned vps would be identical.
Out of interest, how unoptimized would t
Any ideas why a VPS, cloned from another instance (DigitalOcean
"droplets" if it matters), won't run on the new instance?
Everything else is the same; region, memory, disk, hypervisor version etc.
And everything else runs, just not Asterisk, unless I do a make
distclean in the /usr/src/asterisk d
Thanks. I did a while ago, but I couldn't make it "fit" what I wanted to do.
Maybe with my increase Asterisk knowledge now I'll take another look. Thanks!
On 27 November 2016 at 18:27, Richard Mudgett wrote:
> Have you looked into ARI [1]? I think it would be a closer fit to what you
> want to
Thanks, Richard - your code does indeed work reliably 100% of the
time, and thank you for that explanation.
I do think the docs at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SHARED
could do with more clarification.
BTW, there were a couple of typos in your code, so for anyone
variables which
> are kind of global accessible by the channel ID.
> So you might call your Gosub with only the (unique) reference name of the
> variables you wish to pass and then call it from your Gosub.
> -> https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHARED
>
&g
It might be worth pointing out that 1.8x was released 6 years ago,
went into security fix only over 2 years ago, and reached "end of
life/no further fixes" over a year ago.
11.x went into "security fix only" last month - 13 and 14 are the
current versions - can you try with them?
On 23 November
Related to
http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html,
at the moment I'm passing one variable via DIAL.
Now I'd like to pass a whole bunch, and my idea was to rather than
having a great string of
b(synctest3b^setVar^1(something)^2(more things)^3(etc))
and then ge
In the musiconhold.conf example, it says:
announcement=queue-thankyou
;If this option is set for a class, then when callers get put on hold,
the specified sound will be be played to them.
I'm using the "m" option in Dial and was hoping to make use of this feature.
Any dialplan way of getting this
p()
exten => setVar,1,Set(testVar=${ARG1}) ; setter
same => n,Return()
On 3 October 2016 at 23:48, Steve Edwards wrote:
> On Mon, 3 Oct 2016, Jonathan H wrote:
>
>> I've googled and I'm probably missing something pretty newbie 101 here,
>> but is there any way, o
tl;dr Is there ANY way/hack of just telling Asterisk to destroy *all*
WHILE loops it may be nested in at a certain time?
Reason: you know the thing about WHILE loops not only having to have
"seen" their endwhile to finish properly?
If not, a reminder before it gives you 3am sleepless nights:
http
.
Thank you again.
On 9 November 2016 at 12:32, Tony Mountifield wrote:
> In article
> ,
> Jonathan H wrote:
>> Thank you - that makes sense. I've seen something about swapping and
>> optimizing channels on the console, but I didn't realise "optimize"
>&g
's no better way to do those
to things, is there any way to force Asterisk to NOT "optimize" those
channels?
On 9 November 2016 at 00:09, Richard Mudgett wrote:
>
>
> On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H wrote:
>>
>> Asterisk 14.1
>>
>> Here&
Asterisk 14.1
Here's a bit of test dialplan, which works as expected and simulates
exactly what I'm doing at the top of my large dialplan...
[dial-pre-test]
exten => s,1,NoOp()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same => n,Set(LIMIT_WARNING_FILE=time_limit_reached)
same => n,Dia
The extra sound packages for en and fr have ha (home automation) and
wx (weather) broken out into seperate directories, but en_GB doesn't,
although the files seems to exist in the main extra folder.
There's no open ticket about this; I can just "ls" the wx and ha dirs
to a text file and make a lit
Just to say "thank you" on the list, and to confirm that the output of
the command you suggested are as follows:
# ip -6 addr show dev eth0
2: eth0: mtu 1500 state UP qlen 1000
inet6 fe80::601:ddff:fea2:dXX1/64 scope link
valid_lft forever preferred_lft forever
# ip -6 addr show dev e
f
Asterisk tips/gotchas!
On 4 November 2016 at 23:02, John Kiniston wrote:
> Could it be SELinux blocking you?
>
> If you change the path to /tmp does it work?
>
>
> On Fri, Nov 4, 2016 at 3:14 PM, Jonathan H wrote:
>>
>> That's just what I'm using, John.
>&g
n,Verbose(1,Current item is
${hashKey}:${HASH(userPrefs,${hashKey})})
same => n,EndWhile
same => n,Verbose(1,Setting ${prefPairs} to DB)
same => n,Set(DB(userPrefs/${CALLERID(num)})=${prefPairs})
same => n,Hangup()
On 1 November 2016 at 23:29, Joshua Colp wrote:
>
> On Tue, Nov 1,
v6 of course, the VPS
host is set to V6 disabled. and as far as I am aware, and my ITSP doesn't
have IPv6, so I just can't figure out why two IPv4 systems are getting IPv6
"pollution" as it were. And why now??!
Anyway, that's what fixed it for me. Thanks!
On 4 November 20
,,,al,u)=Extension: s)
>
> On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H wrote:
>>
>> Seems I can write to an existing file, but is there really no way of
>> creating a new file to log some da
4 November 2016 at 21:32, John Covici wrote:
> Won't the system command do it?
>
> On Fri, 04 Nov 2016 17:26:13 -0400,
> Jonathan H wrote:
>>
>> Seems I can write to an existing file, but is there really no way of
>> creating a new file to log some data to, withou
Two VPSs. Identical setups with the exception of the extension.
Same version of everything, Asterisk 14.1, Ubuntu 16.10, same firewall
rules and so on - box 2 was cloned from box 1.
Both VPSs run in the same datacentre.
Suddenly, after weeks of OK, I'm getting lots of this on ONE box only:
[No
Seems I can write to an existing file, but is there really no way of
creating a new file to log some data to, without reverting to AGI?
(will be different for each caller ID)
--
_
-- Bandwidth and Colocation Provided by http://ww
All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions.
I don't need any other DB layer, I have no hardware, and I was wondering
what the smallest build possible was.
I experimented, but everything relied on other things. And then I
wondered... is there actually any point? Is
I need to store some basic caller data in ASTDB - certainly doesn't need
full blown mySQL.
There's 4 or 5 bits of info per caller, and I saw that there is a json
entry in ASTDB for the endpoint.
Does that mean that there are accessible functions to deal with json now? I
couldn't find anything in
The first time I run a loop, the AGI returns a list of files.
I append a path from a variable, and play out the files using SHIFT to
loop them.
The FIRST time I enter the system, this is what the complete list to
be looped looks like:
/audio/phone_service_live/menu1-eye_conditions/title,/audio/p
I loop through a list in Asterisk which is generated by a Python AGI
and I've just been bitten by a variable limit I didn't realise existed
before.
The only way I can think of working around this is to get Python to
write the list to file, then do a FILE_COUNT_LINE to get the number of
items (need
Yes! That's the one. Thank you. That's a good workaround.
The following test dialplan shows the bug (feature?)
exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(seconds=57)
same => n,While($[${seconds} <= 400]);
same => n,Set(minutes=$[FLOOR(${seconds} / 60)])
I thought dialplan flow was that (normal!) agi was called, it did its
thing (which include returning some dialplan variables/lists), and
then when agi finished it returned to the dialplan which then reliably
carried the product of agi.
But I'm calling agi, scanning a path in python, and then findi
I'm not mathematically gifted, but shouldn't 957%60 be 15 remainder 57?
Google and my desktop calculator certainly think so.
So where am I going wrong here? The following code
exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(myNum=957)
same => n,Set(sec=$[REMAINDER
me => n(done),Set(CPLAYBACKOFFSET=0)
same => n,Return()
On 18 October 2016 at 17:56, John Kiniston wrote:
> Alright... How about:
>
> exten => 100,1,NoOP()
> same =>
> n,Set(Duration=$[CEIL(${STAT(s,/var/lib/asterisk/moh/reno_project-system.alaw)}
> / 8000)])
>
I can get the size of a ulaw file using STAT.
And I can get the duration in seconds by doing filesize/8000.
Your tea-break challenge is to help me find the shortest most
Asterisk-like way of saying:
"The following file is [ minutes and] seconds long".
...without referring to external applicatio
I'm going to go ahead and file a bug report, 'cos something definitely
ain't right here! Bug filed:
https://issues.asterisk.org/jira/browse/ASTERISK-26481
This bit of dialplan.
exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(featurefile=/home/test/feature-1.txt)
On 17 October 2016 at 16:12, Matt Riddell wrote:
> https://www.npmjs.com/package/speech-rule-engine
Thanks. That and the tip about jackaudio look interesting, although
that thing above is just a parser, not a renderer.
I think, at this stage, it's an idea to go back in the box for another
day.
Lots of little bits in one email to save polluting the list too much today,
time for me to try and give a little back, too!
Someone posted about sngrep a couple of days ago. What a great tool!
Is there a list of useful stuff like this that isn't hopelessly out of date?
Talking about hopelessly ou
r something to navigate the object you create
> and tts to describe current position. The hard part will be parsing the HTML
> even though most HTML is broken :-)
>
> Kind regards,
>
> Matt Riddell
>
> On Oct 17, 2016, at 9:00 AM, Jonathan H wrote:
>
> Has anyone attempte
0,1,l,u)})
same => n,Verbose(1,Using a variable, feature2 is set to
---${feature2}--- and is ${LEN(feature2)} long)
same => n,Set(unfilteredfeat=${FILE(${featurefile},0,1,l,u)})
same => n,Set(feature3=${SHIFT(unfilteredfeat)})
same => n,Verbose(1,Using a string with shift
Has anyone attempted making the web phone accessible? I can only find one
company which operated between 1996 and 2000.
I was thinking, install Chrome with Chromevox, headless, on a server, and
use something like an AGI to send basic keyboard commands to navigate a
page, as a screenreader user wou
I have a plain text file, ASCII, unix line breaks. 1 single line, and all
that is in it is the word "radio".
Here's some test dialplan:
exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN})
same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)})
same => n,Verbose(${feature})
Hmmm, sorry, I can't think of anything except... why do you need the
STUN server? And are you sure that all the ports in your router
definitely match the ones Asterisk thinks it's using?
Then there is always the SIP-ALG problem with some routers, which some
people have been able to overcome by swi
All other things aside, this stands out immediately:
RTT: 434.393 msec
That's almost half a second round trip for a packet. I'm amazed
anything works at all. For SIP connections, mine are usually about
26ms max, anything above about 35 is bad. Looks like a serious config
issue.
Try pinging and s
On 13 October 2016 at 13:18, Tony Mountifield wrote:
> exten => _X,1,NoOp(Matching single digit)
> exten => _X.,1,NoOp(Matching multiple digits)
> exten => _X!,2,SayNumber(${EXTEN})
> exten => _X!,3,Etc..
Thanks - I appreciate the idea, but it matches more than 2 digits.
But, thanks to your inf
rds
>
> Jean Aunis
>
>
> Le 13/10/2016 à 12:54, Jonathan H a écrit :
>>
>> Back to basics here. I want to match on one OR two digits.
>>
>> The following two both work, but ONLY for more than one digit, which
>> is not as expected from the docs (s
Back to basics here. I want to match on one OR two digits.
The following two both work, but ONLY for more than one digit, which
is not as expected from the docs (see below).
exten => _X.,1,SayNumber(${EXTEN})
exten => _[0-9].,1,SayNumber(${EXTEN})
This next one will ONLY match 2 digits, as expe
Are those numbers correct?
Asterisk 12 stopped being supported almost 2 years ago and became "do
not use" on 2015-12-20
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Ubuntu 14 may still be supported, if you're on 14.0.4.5
https://wiki.ubuntu.com/Releases
You could try make distcle
d the
following code. Thank you again.
escape_digits = str("*")
pressed_digit=agi.stream_file(promptFile,escape_digits)
if pressed_digit == "*":
sys.exit(0)
On 11 October 2016 at 09:31, Lefteris Zafiris wrote:
> On Mon, 10 Oct 2016, at 22:47, Jonathan H wrote:
>> For re
For reasons best known to myself, I call a python agi (PYST2 - love
it!) which streams a series of very short files in quick succession.
Like this:
escape_digits = str("0")
agi.stream_file(promptFile,escape_digits)
and this is what I see on the AGI debug:
AGI Tx >> 200 result=0 endpos=6784
AGI
Just a minor thing: on
http://www.asterisk.org/downloads/asterisk/all-asterisk-versions it still
reports 14.0.1 as being the latest version, although the download itself
contains 14.0.2
I'd have file a bug but there doesn't seem to be a "website" section in the
tracker.
On 30 September 2016 at 22
I've got an agi that recognises speech (via Google) and another that turns
text into speech (tts) (via Microsoft Translate).
Both are web APIs, both called via seperate python AGIs.
I've googled and I'm probably missing something pretty newbie 101 here, but
is there any way, or fiddle, that I can
Funnily enough, I was just thinking the same this morning.
I run two boxes, and my idea was the use a sys call to grab the loadav,
multiply that by 1000 and then use that as the delay before answering.
In other words, if box 1 had a loadav of 0.2 and box two have a loadav of
0.5, box 1 would answ
Something like this?
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md
On 27 September 2016 at 19:31, Ryan, Travis wrote:
> So if someone has their own hardware and infrastructure but wants a software
> (not FreePBX but perhaps similar) wha
Thanks, Marcelo. I just subscribed to that bug; you're right, I also
noticed the lack of info when attacking to asterisk. I just forgot to
mention it here!
On 20 September 2016 at 13:43, Marcelo Terres wrote:
> Hello Jonathan,
>
> https://issues.asterisk.org/jira/browse/
Great! Thanks, team, but just before I file a bug..
No matter how many v and d I put, when I now do "dialplan reload" in
v14, it just says "Dialplan reloaded".
Previously, it used to give some info, and I could scroll back and see
if there were any obvious errors in the dialplan.
Is this and int
All your libraries, kernel, headers and build tools up to date?
The other thing that might be worth noting is the warning along the
lines of "contains modules that were not installed by this version of
Asterisk".
Might be worth deleting anything that appears there, and then starting Asterisk.
On
0.0.0:5061
cert_file=/etc/letsencrypt/live/mysite.co.uk/fullchain.pem
priv_key_file=/etc/letsencrypt/live/mysite.co.uk/privkey.pem
method=tlsv1
But this won't be any good to you on sip. What version of Asterisk are
you using?
On 26 August 2016 at 11:36, hw wrote:
> Jonathan H schrieb
Well, what immediately stands out is:
"FILE * open failed!"
Have you triple checked that the full filepath is correct and that the
user that Asterisk is running as has full permissions to access your
valid certificate file?
I have it working with microsip and a free TLS cert from LetsEncrypt.
Whe
Here's a weirdness - I got a call from someone who couldn't get to my info
line earlier, I tried it and it was busy tone.
Being on a layby beside a road on a mobile on a long journey, my only real
option was a remote server reboot so I couldn't diagnose further.
That fixed it, but here's the weir
On 17 August 2016 at 20:40, Jonas Kellens wrote:
> When I compile "--without-pjproject" I loose all webrtc functionality. I get
> errors about the lack of "ice-frag and ice-pwd in the SDP-body".
> So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip).
> Do you have any ot
isk for MOH let me know. The ones that
> I have working is MP3 and MMS.
>
> On Mon, May 9, 2016 at 1:18 PM, Jonathan H wrote:
>
>> Thanks Joshua and everyone,
>>
>> Joshua's solution seems a lot simpler and works well. Only one thing
>> now - The reason I
I'm genuinely fascinated why you are insisting on using a version of
Asterisk almost 3 years old, for which EOL support ended last year.
Is there any particular reason you cannot or will not use the current
version as others have suggested?
Also, I see you are using Doubango and WebRTC, but in th
which was
dialled by the Dial(PJSIP/...) application? Does that make sense?
(It's getting late here!).
Thanks!
On 9 May 2016 at 18:22, Joshua Colp wrote:
> Jonathan H wrote:
>>
>> Thanks Joshua and everyone,
>>
>> Joshua's solution seems a lot simpler and work
m/media/listen.pls
[streamdemo5]
digit=5
mode=custom
application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s
http://206.225.87.121:8000/
On 9 May 2016 at 18:00, A J Stiles wrote:
> On Monday 09 May 2016, Jonathan H wrote:
>> . {stuff deleted} .
>> [streamdemo]
>> exte
-- Executing [s@streamdemo:3]
WaitExten("Local/2@play-radio-0003;1", "") in new stack
-- Timeout on Local/2@play-radio-0003;1, continuing...
On 8 May 2016 at 14:56, Dovid Bender wrote:
> Michael,
>
> What you do is you dial another context and th
I'd like multiple people to be able to dial in and listen to various
live radio streams.
I was told that the correct resource-friendly way would be to setup a
MoH class, and then select that from the dialplan.
This works well, but how do I switch between streams?
Someone correct me if I'm wrong,
>From what I can tell from the Wiki page at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MP3Player,
if someone dials in and starts playing a stream, mp3player will load
up the URL and inject it into the current call.
But what about if 20 or 30 people call in, and it's firing
If it helps, here's a quick n easy guide I made to installing from scratch
with pjsip on Ubuntu.
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md
There are some other scripts like firewall, wizard etc, but there are aimed
at Voipfone users
re "in
sync" with the forum.
Lesson learnt, and again, thank you.
On 18 January 2016 at 11:57, Joshua Colp wrote:
> Jonathan H wrote:
>>
>> Would greatly appreciate any input into this currently-unanswered
>> question on the forum:
>>
>> http://forums.ast
Would greatly appreciate any input into this currently-unanswered
question on the forum:
http://forums.asterisk.org/viewtopic.php?f=1&t=96496
I posted it on Jan 6th, have tried so many things, so much forum/list
searching and late nights since, but have had to admit defeat.
Rather than duplicate
Good evening all.
I am having issues with CDR and confbridge. When the first call is placed into
conference CDR stops tracking time. If I hang the call up the billsec reported
is only up to the the time before the call enters the bridge.
However if a second call joins the bridge the full amount
Does anyone know if it is possible to disable asterisknow-version from writing
over my issues file?
Alternatively is it really required to have it as a dependency in asterisk 13?
Surly everyone has upgraded from the old repository file system and asterisk 13
wasn’t even on the old file system.
Can you write the unique variable to astdb and then write it back to the
variable?
Not sure I have thought this through
J
On 11 Jun 2014 18:42, "Kelly Opal" wrote:
> Hi
> I am trying to set up a hold system so that a call is always
> parked in the same spot no matter how many times it
I didn't know that feature existed.
I'm doing a scripted restart by using the asterisk command line to tell me
how many active calls are current. If 0 then restart.
J
On 28 May 2014 10:52, "Sander Smeenk" wrote:
> Hi,
>
> I want to do a scripted 'restart when convenient' on a daily basis. This
could
> make it work, but for what I'm doing it really is probably the best option
> (especially since it's on a pre-existing Asterisk install that was not
> configured with ODBC support).
>
> -Josh
>
>
>
> On Mon, Apr 21, 2014 at 10:27 AM, Jonathan White wrote:
I’m trying to use the asterisk database but I think there is a limitation in
deleting records I need to make my logic work.
I understand that I can delete all family members with a specific key
and that I can delete an entire family of keys
but I would like to be able to delete specific keys whi
Is there a standard set of prompts to use with ControlPlayback as there are
with many of the other applications like confbridge?
I can’t find a prompt with instructs the user how to use it’s functions.
Perhaps you have to construct a prompt out of multiple prompts. Does anyone
have an idea of w
, 2014 8:02 PM
To: Jonathan White ; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] ControlPlayback can not replay complicated
file names
This doesn't fix the issue, but a work around might be to try using file
names without the any : in them
-Ori
2014, Jonathan White wrote:
If not sure if I am looking at a bug or expected behaviour as I do not see
anything in the documentation.
ControlPlayback can not replay complicated file names
For example it can replay
1005
but it can not replay
1005-2014-04-08_23:58:17
On Thu, 10 Apr 2014, Eric Wi
If not sure if I am looking at a bug or expected behaviour as I do not see
anything in the documentation.
ControlPlayback can not replay complicated file names
For example it can replay
1005
but it can not replay
1005-2014-04-08_23:58:17
Playback can replay
1005-2014-04-08_23:58:17
I suspect t
motty cruz wrote:
> Hello Jonathan,
> I thank you for prompt reply to my post.
>
> I'm using SIP trunks with Polycom sp450 devices.
>
>
> Also, I was wrong to mention meetme, my conference does not involve
> using meetme feature on Asterisk.
>
>
> It do
that
your problem will be fixed if you upgrade. r373242 comes to mind in
particular.
Other than that though, it would be helpful if you added some
additional information, such as what arguments are are running meetme
with and what kinds of devices you are connecting with (SIP phones
presumably?)
--
J
u can't get the required libraries for CentOS5?
Without knowing what the differences are between libs you have on the
CentOS5 and CentOS6 machines... I really don't have a lot to go on.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 J
Jerry Geis wrote:
> I have a CentOS 6.3 machine I installed Asterisk 11, worked fine...
>
> I then tried to install on Cents 5.8, seemed to go fine... Then when
> I
> placed a call I got this:
> ast_rtp_instance_new: No RTP engine was found. Do you have one
> loaded?
>
> Did a search and found is
eeing. If the device
has a static IP and you don't want to deal with registration, you
could always change the host to that IP address. Alternatively you
could just figure out how to get your devices to register to your
Asterisk server.
--
Jonathan R. Rose
Digium, Inc. | Software Eng
Harish Mandowara wrote:
> I have Asterisk server 1.8.19 with jabber enabled.
>
> On the other side i have openfire server with asterisk-im enabled.
>
> I have a doubt, whether my sip client connected with asterisk can
> send message to other sip client, which is connected to same
> asterisk serv
Jonas Kellens wrote:
> Hello,
>
> using Asterisk 1.8.12.2
I think that was tagged before any of my recent app_queue patches. In that case,
it might work if you just update to the latest 1.8 release. If it doesn't, go
ahead and
file an issue on JIRA.
--
Jonathan R. Rose
Digium, I
I was poking around with the Add/Remove QueueMember code a while back. From
the sound of what you are saying I might have just missed something critical.
for your case.
It'd be good to know what version you are using so that I can verify whether or
not my changes could have affected you.
wn to you in a nice easy to read view. But that sort
of thing might take a bit of work.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139
Check us out at: http://digium.com & http://asterisk.org
--
2 for whatever reason, then you can use 1.8 on a test system to
see if you can reproduce
your problem and if you can, file a bug report against that and hope the patch
either translates
well to 1.6.2 without much intervention or you could attempt to backport it
yourself if it doesn't.
Goo
pproach though would be that
MixMonitor will automatically mix audio from both ends of the call
into a single recording. That behavior can be worked around starting
with Asterisk 10 by using the r and t options.
I guess it's worth noting that if you aren't using 1.8 or higher
there isn&#
Jonathan Rose wrote:
> Sean Darcy wrote:
>
> > I'm building asterisk 11 beta 2. I've been using silk a lot. I
> > don't
> > see
> > silk listed in menuselect as a codec. But I also don't see an
> > asterisk
> > 11 silk cod
rather than 11, but the
architecture for codec translators remains largely unchanged, so I would guess
it'll probably work. Probably.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 42
complete confbridge (type 'confbridge',
don't hit enter but do hit tab once or twice and see what gets listed).
I hope that helps.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive
1.8 is EOL?
Last I checked, not until 2015-10-21.
I'm sure you are probably already of the following, but for anyone else who
reads that message and is confused...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Asterisk 11 is just the next LTS release after 1.8. The new ver
e of
the major features you might just consider looking at the Asterisk
wiki.
https://wiki.asterisk.org/wiki/display/AST/New+in+10
https://wiki.asterisk.org/wiki/display/AST/New+in+11
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
dire
f it's the case that you are calling people get getting a busy
signal or something, you might consider sending them to some
extension that verifies that they answered before sending them
into the conference or something similar to that.
--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445
Thorben Jensen wrote:
>> I was looking over Queue and I don't think there is actually an
>> option for Queue that will automatically start a MixMonitor. I see a
>> few options
>> involving mixmonitor (x and X), but they appear to be more about
>> allowing
>> the parties involved with the call to
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