Sangoma purchased Digium.
You can find Sangoma cards at https://www.sangoma.com/telephony-cards/
On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad wrote:
> Hello All;
>
> We were using Digium cards, now I am not able to reach for digium website
> that contains the telephony cards and Asterisk websi
Hello,
I'm working on converting my 18.0.1 test system from SIP to PJSIP and I've
run into something odd.
I have a queue defined named acme-test that has two agents in it,
PJSIP/7001acme and PJSIP/7002acme.
I have autohints=yes in my acme-intern context, I have not defined hints
for either of th
You could do the old school method and create and move a .call file from
your dialplan.
exten => writefile,1,NoOP()
same => n,Set(CALLFILE=/var/spool/asterisk/tmp/${FileName}-${ARG1}.call)
same => n,Set(FILE(${CALLFILE},,,al,u)=Channel: SIP/bob)
same => n,Set(FILE(${CALLFILE},,,al,u)=WaitTime:
r docs for *_reply_codes modparam)
> to accept a 404 reply to a SIP:OPTIONS as a valid response.
>
>
> Hope it helps.
>
> Cheers,
> Joel.
>
>
> On Thu, Jul 16, 2020 at 5:04 PM John Kiniston
> wrote:
>
>> I'm implementing a SBC with my Asterisk PBX
I'm implementing a SBC with my Asterisk PBX but the keeps disabling the
trunk group I've configured and I think it may be because Asterisk is
returning a 4r04 to the OPTIONS.
I've created a test context and have put in a wildcard pattern match to try
and catch those options but it doesn't seem to
Dovid, You could use func_odb + a ODBC Redis driver to keep from having to
shell out.
On Wed, Jul 8, 2020 at 4:37 AM Dovid Bender wrote:
> Hi,
>
> Does anyone know of any projects that would allow you to use Redis in
> place of AstDB? By in place of I don't mean for what Asterisk needs but to
>
Nice, Do you have the code up on GitHub? I'd love to see it.
What's the source of the data? Something API driven I hope?
Have you thought about implementing your project via curl instead of
func_odbc?
On Wed, May 27, 2020, 8:52 PM Saint Michael wrote:
> In a few weeks, no SIP call is going to
Use the ARRAY version of Set.
same = n,ExecIf($["A" = "B"]?Set(ARRAY(C,D)=1,2))
On Tue, Apr 21, 2020 at 3:56 AM Administrator wrote:
> Hello,
>
> we want to use something like
>
> same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...)
>
> Problem is that result gives C=1) & Set(D=2) & ...
>
>
rom looking at the wiki won't STRFIME just give me what I need based on
> the unix time that I put in? What I am actually looking to do is convert
> over from 12 hour format to 24 (unless strftime does just that and I don't
> kow what am I am doing?).
>
>
>
> O
Try using the STRFIME function instead of doing this by hand.
https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
*%H*
The hour as a decimal number using a 24-hour clock (range 00 to 23).
*%I*
The hour as a decimal number using a 12-hour clock (range 01 to 12).
On Thu, Feb 13, 2020 a
Ira,
What version of Asterisk are you using, and what channel driver?
There has to be a better way than to create hundreds of peer entries.
On Thu, Dec 12, 2019 at 12:26 PM Ira wrote:
> Hello Jan,
>
> Tuesday, December 3, 2019, 8:49:28 PM, you wrote:
>
> Jan> The next thing to look at is firew
Jerry, What if you specify a higher bitrate to mpg123?
You are limiting it to 8k with the 'r' option.
I convert my source audio files with sox to 16khz signed linear for
wideband hold music.
sox -c1 hold.wav -r 16000 -c 1 -e signed-integer -r 16k hold.raw
Then I rename the .raw file to a .sln16
On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote:
> Anyone know how someone can hack an asterisk box and register with every
> single account on the box.
>
> This box only has 3 accounts, with very complex passwords. Have VoIP
> blacklist setup and fail2ban…
>
I've seen this happen when web
The solution you don't want to use (testing the result of Read) is what I
personally use.
same => n(PROMPT),Read(RESPONSE,playback_or_callback,1,,3,8)
same => n,GotoIf($["${RESPONSE}" = "1"]?RETURN,1)
same => n,GotoIf($["${RESPONSE}" = "2"]?PLAY,1:PROMPT)
On Mon, Apr 8, 2019 at 11:45 AM Dovid
Thomas,
Does the Asterisk box need to do anything other than handle calls for this
one specific IVR? IE does it ever originate calls?
If it's only recieving calls then I'd turn on guest access and not even
bother with a peer.
Just set
[general]
context=transit-ivr
allowguest=yes
On Tue, Feb 26
ifferent discussion.
On Wed, Feb 20, 2019 at 12:14 PM Brian J. Murrell
wrote:
> On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote:
> > Use the IF function to evaluate and change the dial command directly.
>
> Thanks for taking the time, but that doesn't actuall
Use the IF function to evaluate and change the dial command directly.
My braces and parens may be off in this example sorry if it doesn't work
out of the box.
exten => s,n,Dial(${IF($["${SIP}" = "PJSIP"]?
${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})}{ARG2})},20,TtWw)
On Wed, Feb 20, 201
How is your endpoint currently configured in asterisk? Have you tried
rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk
can send audio back to the client?
Alternatively if your SIP Proxy is also a Media proxy you could set the
media_address on the endpoint to be your proxy
Davor,
Have you attempted to do a Set(PJSIP_HEADER(remove,Diversion)=)) in your
add_diversion context to remove the header Asterisk is passing through?
On Mon, Dec 10, 2018 at 3:04 AM Davor Jovanovic
wrote:
> Hi all,
>
> I’m trying to rewrite Diversion header when call forwarding is done on
> t
This should work, How are you defining your timeouts in the queues.conf ?
And to verify, in your extensions.conf you are calling Queue with the queue
name and the ruleset to apply from queuerules.conf?
On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote:
> Hi All
>
> I have been looking at this
o.
>
> Also, my queues.conf setup like this:
>
> timeout=30
> retry=1
>
> Which means if I send it to "Eric" - it will go to his voicemail after 30
> seconds. Should I change timings?
>
> Thank you!
>
> --
> *From:* John Kiniston
> *To:* Ivan Demkovit
; What do you mean by using LOCAL channel? Can you be more specific? I'm not
> very good at this :)
>
>
>
> This is logger.conf. Where(which section) should I place logging
> configuration?
>
> [general]
> dateformat=%F %T
>
> [logfiles]
> console => n
what does the output of 'queue show sales' show?
Do you have queue logging enabled? Have you looked in the queue log to see
what events are firing?
On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch
wrote:
> Hello,
>
> I have queues.conf setup with a group like so:
>
> [Sales](StandardQueue)
> ann
Good Morning John,
Most likely you would need to make changes on your phones themselves.
PJSIP is just sending NOTIFY messages when there is a state change, The
phone itself is choosing how to handle the message.
On Mon, Nov 5, 2018 at 10:40 AM John T. Bittner wrote:
> Anyone know how to turn
You could use GROUP & GROUP_COUNT to track how many channels you are using
before you attempt to dial out and send back a Busy/Congestion/Whatever to
your endpoint when you are at your limit.
On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin wrote:
> Hello,
>
> I am running Asterisk 11.17 with DAHDI
It's a configuration issue with the peer (or the peer it's matching) in
sip.conf
You have configured the peer to use avpf but your phone is not attempting
to do avpf.
Either configure the phone to match the entry in sip.conf or change
sip.conf to match the phone.
On Mon, Aug 27, 2018 at 4:56 PM
I use sudo to limit this.
Cmnd_Alias CAPTAGENT = /sbin/service captagent stop, /sbin/service
captagent start, /sbin/service captagent restart
Cmnd_Alias ASTERISK = /sbin/service asterisk stop, /sbin/service asterisk
start, /sbin/service asterisk restart, /usr/sbin/rasterisk,
/usr/sbin/asterisk, /u
On Fri, Jul 20, 2018 at 11:41 AM Saint Michael wrote:
> The community would benefit if a non/licensed version of G729 would be
>> included with Asterisk, since the license expired. The current codec
>> source code posted still requires licensing.
>>
> I am sure Digium would not prefer to
>
>
David,
You should be able to use the Bridge dialplan application to do what you
want.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge
I use the CHANNELS function and the IMPORT function to find the channel to
bridge to my caller.
On Sun, Jul 8, 2018 at 8:17 PM David C
I don't believe it's supported to start an audio call and then re-invite to
add video, I believe Asterisk requires the call to start as a video call.
On Mon, Jun 11, 2018 at 6:42 AM, Stefan Tichy wrote:
> Hi,
>
> when some phone inititates an audio call and sends a re-invite with
> audio and vid
I got excited when I saw 8 new messages on the Asterisk list-serve this
morning, What discussions must be happening I thought!
You are a tease sir.
On Tue, May 22, 2018 at 7:58 PM, Matt Fredrickson
wrote:
> More testing. Test test test. :-)
>
> --
> Matthew Fredrickson
> Digium, Inc. | Enginee
Lock files.
Create one when you start sending the fax, on your retry process check for
a lock file and if one exists don't retry.
On Mon, May 21, 2018 at 10:49 AM, D'Arcy Cain
wrote:
> I am having troubles with sending faxes. I hope someone can help me
> work out a better method.
>
> Basically
On Tue, Apr 24, 2018 at 9:56 AM, Bruce Ferrell
wrote:
I'd REALLY like to get it working. And for the record, I REALLY HATE pjsip.
>
> I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux
> since '93 and telecom since 1980. The config is so opaque, poorly
> documented and erro
I'd suggest doing packet captures on the T21P's themselves at the affected
branches and see if you can catch it happening.
The Yealinks themselves will regenerate DTMF if they get signaled for it.
On Thu, Feb 8, 2018 at 2:19 AM, Stefan Viljoen
wrote:
> Hi Guys
>
>
>
> I’ve got a situation where
use func_odbc, create a new function that does a lookup.
[CALLERID]
prefix=LOOKUP
dsn=MyDB
readsql=SELECT CALLERID from MyNames where CallerIdNum =
'${SQL_ESC(${ARG1})}'
exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${CALLERID(NUM)}))
On Wed, Jan 17, 2018 at 6:16 AM, Atux Atux wrote:
> Hi.
sterisk+11+Function_DB
On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux wrote:
> That is the general idea. But how do i make it work? is there somewhere
> ready?
>
>
> On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston
> wrote:
>
>> Define your *72 and *73 extensions in your internal
Define your *72 and *73 extensions in your internal context, Have them set
a value in the ASTDB that you then check when dialing your handsets.
The same can be done for call forwarding, store a number in the ASTDB and
check if it's present, if it is forward the call to that number.
On Wed, Jan 10
application ? Using externnotify ?
> 2. What is MWI:101@default expression for (see [2] ?
>
> Cheers
>
> [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+
> button+to+Monitor+a+Voicemail+Box
>
> 2017-11-21 17:58 GMT+01:00 John Kiniston :
>
>> Hello Olivier,
Hello Olivier,
I may be incorrect but I don't believe you can hint on a mailbox like that.
I've always used custom device states and dialplan logic for my shared
voicemail boxes that are not being watched directly by a endpoint natively.
On Mon, Nov 20, 2017 at 5:07 AM, Olivier wrote:
> Hello,
The easiest thing to do might be to limit the peer to 11 calls in your
sip.conf or pjsip.conf and then use the GROUP and GROUP_COUNT functions to
limit the extension to 10 concurrent calls, Then when you get call #11 who
is high priority you can allow it through or otherwise return busy.
https://w
I'm toying with the idea of replacing a statically generated file I include
in my extensions.conf with a realtime lookup against my database.
I've got it working but something seems off in my logs, It looks like I'm
getting two lookups for every priority?
[Oct 12 16:45:24] DEBUG[26541][C-000e
Well, I can answer one of your questions.
To see the current external_media_address and external_signalling_address
from the CLI you can issue a 'pjsip show transport '
Your router could have a built in sip helper that is rewriting the contact
for your packets.
On Mon, Oct 9, 2017 at 2:56 PM, O.
Yes, You could do easily this either with the internal asterisk database or
with something like func_odbc as a source for the data.
In the context you receive your incoming calls you do a lookup against one
of the above data sources using the CALLERID(NUM) and change CALLERID(NAME)
to be the name
My only suggestion would be you could reduce your line count by replacing
your GotoIf statements with ExecIF statements.
exten => addheader,1,ExecIf($["x${ARG1}" !=
"x"]?Set(PJSIP_HEADER(add,Route)=${ARG1}))
same => n,ExecIf($["x${ARG2}" !=
"x"]?Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2}))
I'm messing around with pre-dialer handlers today and running into a wall.
Dial has the U option where I can execute a Gosub when the channels bridge
and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like
the channel I called was Busy.
I want to do something similar with a Pr
You can do this with cdr_adaptive_odbc.
On Tue, Jun 20, 2017 at 5:58 AM, Marcelo Terres wrote:
> Well, you could create and AGI and run it after the normal CDR was
> inserted.
> Marcelo H. Terres
> IM: mhter...@jabber.mundoopensource.com.br
> https://www.mundoopensource.com.br
> https://twitter
I have a real ugly queue that has this in it's rules
[mrule]
penaltychange => 20,2,2
penaltychange => 40,3,3
penaltychange => 80,4,4
penaltychange => 120,5,5
penaltychange => 150,6,6
penaltychange => 180,1,1
penaltychange => 200,2,2
penaltychange => 220,3,3
penaltychange => 240,4,4
penaltychange =
You could use the DIALGROUP function for this and not need to shell out.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_DIALGROUP
On Mon, May 8, 2017 at 7:35 AM, Frank Vanoni
wrote:
> Hello
>
> I have the following scenario:
>
> [mynicecontext]
> exten => 2000,1,Dial(SIP/device
Well, My suggestion was to use FUNC_ODBC but instead you went with
APP_MYSQL which has been depricated.
Did you compile APP_MYSQL? It's not enabled by default.
On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux wrote:
> Thanks a lot for the reply.
> I did follow that already, but i do have a problem. H
You can use func_odbc to do this.
https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2
There is a good chapter in the Asterisk book about using ODBC for
hotdesking that may help you understand ODBC as well.
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-b
type=peer matches on the IP of the specified host, If you want to match on
the username use type=user.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: http
Your new file is in the 'myfolder/1'' subdirectory of the MOH directory.
Either move the file into the MOH directory or define a new class in
musiconhold.conf that is for your directory.
On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens
wrote:
> Hello
>
> using Asterisk 1.8.32.3
>
> Current music
'sip show settings' may do what you want.
On Wed, Jan 11, 2017 at 3:32 AM, Thufir Hawat
wrote:
> I appreciate that the console lets you see the details for a peer with
> "sip show peer foo". Certainly, I can look in sip.conf to see the
> [general] context, but can I output those settings, and o
On Mon, Jan 2, 2017 at 5:26 AM, Thufir Hawat wrote:
>
> But, their registration string with Asterisk is:
>
> Locate [general] secion and add the following
> register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010
>
>
> Wouldn't this send every outbound call through that Anveo account?
>
>
>
R
You could use the IAXVAR() function to set variables before dialing your
IAX peer on the initial PBX that then get retrieved by the 2nd PBX.
PBXA:
same => n,Set(IAXVAR(CALLERID)=${CALLERID(num)})
same => n,Set(IAXVAR(DNID)=${CALLERID(DNID)})
PBXB:
same => n,NoOP(CallerID is ${IAXVAR(CALLERID)} DNI
Line 161 of include/asterisk/channel.h:#define AST_MAX_ACCOUNT_CODE20
/*!< Max length of an account code */
That's where your limit is coming from, I see some other places where the
code doesn't use that definition ( chan_ooh323, cdr_sqlite, cdr_tds ,
res_config_sqlite) So as long as you are n
mat:
> Cannot open '/home/logs/anonymous.txt': No such file or directory
> [Nov 4 21:46:16] ERROR[1676][C-0003]: func_env.c:949 file_write:
> File '/home/logs/anonymous.txt' not in line format
>
> Asterisk is running as root (yeah, I know!), and has permissions on
I'm able to use the FILE function to create files just fine.
Set(FILE(${CALLFILE},,,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 data to, without reverting to AGI?
>
I always set a TIMEOUT(absolute) on calls across trunks to something
reasonable like 10 hours, that way calls should end in a sane amount of
time even if something weird happens.
Otherwise I've always had to do a reload when I couldn't hang up from the
CLI.
On Thu, Nov 3, 2016 at 9:16 AM, Carlos
OOTAI
wrote:
> Le 24/10/2016 à 18:46, John Kiniston a écrit :
>
>> You can do it with IAXVAR.
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_IAXVAR
>>
>>
>> exten => 100,1,Set(IAXVAR(myvar)=foo)
>> exten => 100,n,Dial(OT
You can do it with IAXVAR.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_IAXVAR
exten => 100,1,Set(IAXVAR(myvar)=foo)
exten => 100,n,Dial(OTHERHOST/201)
on OTHERHOST
exten => 201,1,Set(myvar=${IAXVAR(myvar)})
exten => 201,n,NoOP(My variable is ${myvar})
On Mon, Oct 24, 2016 a
This is the most compact I can make it just by tidying up your variables
and playbacks:
same =>
n,Set(ARRAY(minSpeech,playFile)=minutes&and,/var/lib/asterisk/sounds/en/abandon-all-hope)
same => n,Gosub(setup)
same => n,set(playFile=/tmp/reno_project-system)
same => n,Gosub(setup)
Alright... How about:
exten => 100,1,NoOP()
same =>
n,Set(Duration=$[CEIL(${STAT(s,/var/lib/asterisk/moh/reno_project-system.alaw)}
/ 8000)])
same => n,NoOP(Duration is $[FLOOR(${Duration} / 60)] Minutes,
$[REMAINDER(${Duration},60)] Seconds)
same => n,Hangup()
On Tue, Oct 18, 2016
I'm trying to find where you configure the parking lot used by phones
registered via pjsip.
In sip.conf you could set the default lot for call parking with the
'parkinglot=mylot' setting but I don't see an equivalent in pjsip.conf
Do I need to use setvar to set CHANNEL(parkinglot) on my endpoint
I'm working on my sip to pjsip translation.
Right now I do some functionality based on what the user agent is on the
calling phone using:
${SIPPEER(${CHANNEL(peername)},useragent)}
I'm trying to replace it with
PJSIP_CONTACT(${CHANNEL(contact)},user-agent}) but I'm not getting any data
returned
I'd recommend trying the support resources for the GUI that is managing
your asterisk installation.
This looks like it might be FreePBX dialplan logic to me, Most likely it
won't be something that the list will be able to help you modify.
http://community.freepbx.org/ is the FreePBX Community sup
Here is a quick and dirty bash script to do it that I wrote you.
#!/bin/bash
if ( asterisk -rx "database deltree blacklist")
then
echo "Blacklist Cleared"
else
err "ERROR Failed to clear Blacklist, Exiting."
exit 1;
fi
while IFS=, re
On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis wrote:
> Sorry, I forgot to write that the SIP peer must keep ringing while the
> announcement is being played.
>
> Le 22/08/2016 à 17:42, John Kiniston a écrit :
>
> This seems like the obvious answer but maybe I'm misundersta
You could try using RetryDial() instead of Dial, It supports playing an
announcement.
On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis wrote:
> Sorry, I forgot to write that the SIP peer must keep ringing while the
> announcement is being played.
>
> Le 22/08/2016 à 17:42, John Kini
This seems like the obvious answer but maybe I'm misunderstanding the
question.
exten => s,1,Dial(SIP/alice,20)
same => n,Playback(myannouncement)
same => n,NoOP(Whatever else you want to do goes here)
On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis wrote:
> Hello,
>
> I am searching a way to
You can delete them from the astdb with database del.
do a 'database show' and your devices should show up in the tree under
'CustomDevstate'
On Mon, Aug 15, 2016 at 9:42 AM, Tomas Holy wrote:
> Hello list members,
> after programing of dialplan I have some messy Custom:hints which I can
> see
I think you almost have it.
In your vmfwd context have a wildcard match that sends the caller back to
the originating voicemail and then define specific extensions that are
allowed to forward.
[vmfwd]
exten => _,1,Voicemail(box@context,option)
same => n,Hangup
; Andrew Ruthven
ext
If you just need the name of the system it may be contained in the variable
${SYSTEMNAME}.
This is assuming you have the systemname set in asterisk.conf
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Main+Configuration+File
That said, for SHELL support you probably need to set :
live_dange
> ...
> -- User entered nothing.
> -- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-04b2",
> "1?REJECT,1") in new stack
>
> Any idea?
>
>
>
>
>
>
> 2016-06-30 21:50 GMT+02:00 John Kiniston :
>
>> I think a simpler
I think a simpler way to do this would be to define an member in your
queues.conf that points to a local channel that calls the remote users cell
phone.
You can use the M option in your dial to run a macro to prompt the user to
accept the call.
Here's my connector macro, I call it with:
Dial(LOC
Use Local Channels and hints to combine SIP/MOM and SIP/MOMMobile into a
single extension you add to the queue.
extensions.conf:
[queue-phones]
exten => MOM,1,Dial(SIP/MOM&SIP/MOMMOBILE,60,tkw)
exten => MOM,hint,SIP/MOM&SIP/MOMMOBILE
exten => DAD,1,Dial(SIP/DAD&SIP/DADMOBILE,60,tkw)
exten => DA
You could explore using ARI with it's Push configuration.
https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration
On Mon, May 16, 2016 at 11:33 AM, Goke Aruna wrote:
> hi all,
> can anyone give me a guide on any asterisk admin solution / interface for
> config management, and monitori
Howdy everyone,
I'm writing a little click to dial type tool and I've run into a snag where
my Originate command needs to call a Sub routine to do a database lookup
and some other stuff.
I can't seem to get the syntax right to call Gosub with Originate
Just testing with the command line I've bee
Have you tried using the table definition that comes with the Asterisk
source?
the file mysql_config.sql is located in contrib/realtime/mysql and defines
a very different voicemail table than what you have in your configuration.
On Tue, May 3, 2016 at 3:10 AM, Michele Pinassi
wrote:
> Hi all,
>
Could you setup a local instance of Icecast and point your PBX to it?
It's been years since I did any streaming but I recall my icecast relay
would only consume bandwith when it had listeners connected to it.
Then you wouldn't have to worry about how many people were listening to a
single channel
You can do this with setting up an application map using DYNAMIC_FEATURES
and enabling it on your incoming call paths.
https://wiki.asterisk.org/wiki/display/AST/Custom+Dynamic+Features
If you don't want to even answer the calls you could try doing this with
'ex-girlfriend logic', I personally ha
I think I saw an old page on the voip-info wiki on how to use CMU Sphinx
with asterisk.
http://www.voip-info.org/wiki/view/Sphinx
IMHO It's not going to be anywhere as good as a commercial product without
a lot of work.
On Mon, Feb 22, 2016 at 11:34 AM, Daniel Chavez
wrote:
> Thanks for the li
I saw Lumenvox offering Speech Recognition for asterisk at a past Astricon.
http://www.lumenvox.com/partners/digium/Asterisk.aspx
On Mon, Feb 22, 2016 at 11:00 AM, Daniel Chavez
wrote:
> Hello list,
> I was wondering if it were possible for asterisk to do a voice recognition
> type IVR?
> Like
You could use Custom Device States to create your 100 extensions to watch
and update their status by changing the states manually.
With Asterisk 13.0 and PJSip I had the RLS feature working with some
Yealink phones (T28P and a T41P to be specific), on the phone I set:
phone_setting.auto_blf_list
On Tue, Feb 2, 2016 at 11:11 AM, Richard Mudgett
wrote:
>
> Since you didn't specify the channel driver, I took a quick look at the
> chan_dahdi, chan_sip, and chan_pjsip channel drivers to see if they
> set any default groups. I didn't see any of those channel drivers set
> default pickup group
Should setting a namedcallgroup & namedpickupgroup supersede numeric
callgroups and pickupgroup ?
I've got 5 peers on my 13.7.0 box,
Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and
Two of them have a namedcallgroup & namedpickupgroup of 'sanday'.
I'm not specifying a num
Just an idea for a work around, Have you thought about putting a proxy
between your PBX and the Internet such as openSIPS or Kamilio?
That way you may not need to change your IP inside pjsip, Let your proxy
handle it.
I gave up switching my edge asterisk to pjsip at least twice because I
> could
On my older Asterisk installs I'm still using Automon because I can set
MONITOR_EXEC to run my post process command and use MONITOR_EXEC_ARGS to
send it some options I need by adding those to my sip.conf entries with
SetVar lines.
On my Asterisk 13.7.0 box I want to use the
recordonfeature/recordo
Have you turned on sip debugging?
Do you see the caller ID in the invite from your Gateway to your PBX?
On Tue, Jan 26, 2016 at 2:07 AM, Belal
wrote:
> Dear sir,
>
> what about receiving call from a GSM gateway. I didn't see the caller ID?.
> is it happen to you? and what is the solution,Please
Have you checked your indications.conf? I've seen a missing or
misconfiguration in the zone definition cause this.
On Tue, Nov 3, 2015 at 11:07 AM, sean darcy wrote:
> On 11/01/2015 12:38 PM, sean darcy wrote:
>
>> I'm not getting any ringing when I use option r with Dial:
>>
>> Dial("DAHDI/1-1"
n Tue, Jul 28, 2015 at 8:51 PM, Andrew Martin wrote:
> - Original Message -
> > From: "John Kiniston"
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Tuesday, July 28, 2015
In your queues.conf do you have a leavewhenempty and joinempty set?
in queues.conf
[myqueue]
leavewhenempty = strict
joinempty = strict
strategy = ringall
ringinuse = no
On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin wrote:
> Hello,
>
> I am running Asterisk 11 on CentOS 6.x. I have configured
Deliver your voicemail to two boxes, One set to email the attached file and
other set not to attach.
exten => 700,1,Voicemail(701@users&702&users,u)
[users]
701 => 1234,Hello There,he...@how.are.you,,attach=no
702 => 1234,Hi Hi,he...@how.are.you,,attach=yes|delete=yes
--
__
I don't see that the Authenticate application has return values for failure
cases or returns call control on a failure case.
Sorry I don't think you will be able to do what you want with it.
On Tue, Jul 7, 2015 at 12:22 PM, Motty Cruz wrote:
> Here is what i have,
> exten => _011X,1,Se
Nice!
I didn't know what dialing rules may apply to his location, Your code does
look like an improvement on mine tho.
I love the REGEX function.
Even better, if the first 4 digits are "0049", you could replace them with
> "0"
> as though it was an inland call:
>
> ExecIf(REGEX("^0049."
> ${CALL
The Authenticate application will do this for you.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate
You can either give it a single PIN to use for all calls, Authenticate
using a value in the Asterisk Database, Or use a plain text file for the
PIN's
On Mon, Jul
The easiest solution may be to strip the leading zero's off your caller ID
before your caller enters the Voicemail app to leave you a message.
ExecIf(REGEX("^[0][0]."
${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2}))
On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello
wrote:
> Hi list!
>
Try this for CHAN_SIP:
same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox
same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
If you are using PJSIP it's more complex
You can use a custom device state to do it.
[dnd]
;DND Toggle
exten => *363,1,Answer()
same =>
n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})})
same => n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1)
;DND On
exten => *78,1,NoOP(Turning DND On)
same => n,Se
BABY appears to be a global variable in your example.
In your CLI output testcarrier is a peer, It's not a variable at all.
The context field for your peer testcarrier is where incoming calls from
testcarrirer will be routed to.
Here is some example dialplan showing how you can use one context
'
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