Hi all
i have run into a problem and cant seem to find the solution
calls that are recorded and lots of voicemails recorded you can her some of
the words repeated as if the person has said it twice
it happens by different callers
using pjsip on 18.9.0
any ideas?
thanks
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On Mon, Jan 17, 2022, 19:58 wrote:
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Hi all
Does anyone have working settings for a audiocodes fxs gateway behind a
firewall to send faxes
thru asterisk not behind nat
i have tried multiple settings and haven't gotten it to work even partially
thanks,
israel
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well looks likes we solved it
the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk
sends a rtpkeepalive a cn packet is sent
the same time a cn packet is sent asterisk loses the dtmf it was sent
On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb wrote:
> Hi all
>
Hi all
i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and
then calls a second server (leg B) server B is a freeswitch server
the servers are configured all thru with rfc2833 for dtmf
the caller enters a number a long 15 digit number like a credit card number
or even a phone
Hi Guys
we have a system that uses a lot of custom hints based on the extension
the extensions use the format of ext-system for example 200-pbx01
when starting asterisk the "core show hints" show the correct hints and blf
works as expected
in the extensions.conf we have _.,hint,Custom:${exten}
-28780
i do see alot of asterisk notices in asterisk 16 alot
translate.c: 12547 lost frame(s) 12548/0 (slin@8000)->(alaw@8000)
On Tuesday 12 May 2020 at 12:28:51, Israel Gottlieb wrote:
>* Hi guys i upgraded to asterisk 16.10
*
>From what? Did you change anything else at the
Hi guys i upgraded to asterisk 16.10 and in most recordings you here only
leg A in the recording
sometimes you might hear a word of leg B
Did any body hit this problem?
Thanks,
israel
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hi all
im trying to call a door phone supporting video
i hear the audio but dont get video
i see this in the log
why should it try to translate?
Unable to find a codec translation path: (h264) -> (opus)
asterisk version 13.26
thanks for any help
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does anyone know how i could use codec opus with asterisk 16 when using
centos 6
the prebuilt binary from digium doesnt load
Thanks,
Israel
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look at zoiper
oem.zoiper.com
you could create a url that creates a build with all credentials
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hi all
How could conferance in a 3rd caller without put the second caller on hold
i would like to press a feature code mid call and have a 3rd caller enter
the call
this could be a real person or a automated system to take credit card info
mid call
thanks,
Israel
--
how about sticking in a pbx between [c] and [h]
so when [h] hangsup you send to [s] if that is 3rd party else i dont see
how you could redirect [c] at all
else maybe ask them to have [h] redirect [c] to [s] then [h] will also be
out of the call
On Mon, Jul 1, 2019, 20:03 Send asterisk-users
Does he have the same voicemail context?
Maybe your firewall is blocking receiving packets from that provider or some
sip helper is messing the returning packets so asterisk is not recieving a
response and resending the invite
Original Message
From: j...@jeff.net
Sent: February 22, 2017 7:57 PM
To: asterisk-users@lists.digium.com
Disable all sip alg/helpers in the router
Original Message
From: andregronwal...@gmail.com
Sent: February 13, 2017 6:40 PM
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] First SIP-registering succeeds, second doesn't
Some further
there are providers which let you call directly to voicemail by using a
prefix
On Mon, Feb 6, 2017 at 8:28 PM, Tech Support
wrote:
> I remember doing the testing and two calls going out at the same time
> don’t actually have to go out at the *exact* same time. The
snom could get lots of configuration options thru sip notify
i once tried updateing the display name on hot desking but ran in to his
problem of having to add it to sip conf staticly
On Wed, Jan 18, 2017 at 5:13 PM, Mark Wiater
wrote:
>
> On 1/18/2017 9:58 AM, Tech
Why not just timing test
It shows the timer used
On Nov 16, 2016 8:13 AM, "Stefan Viljoen" wrote:
> Date: Tue, 15 Nov 2016 17:52:07 +0100
> From: Olivier
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
Are you sending progress?
בתאריך 24 באוג׳ 2016 13:40, "Saint Michael" כתב:
> I have the same exact issue. I cannot push any sounds or even Playtones
> to the caller, unless the channel is answered, which is not possible for
> billing reasons.
> I am also using the Local
e Playback means that SIP/alice should continue
> to ring for the remaining 20 of the 40 seconds, as the Playback will not
> answer (terminate) the call.
>
> Don't forget AstriCon this year - www.astricon.net
>
> On 23 August 2016 at 12:52, Israel Gottlieb <isr...@gmail.com&g
You could m and make a moh file that has ringing the first 30 sec and then
the anouncment
בתאריך 22 באוג׳ 2016 7:19 PM, "Jean Aunis" כתב:
> Thank you for the idea. The problem with RetryDial, is that it will cancel
> the first call, play the announce and then dial the
Could you please write the problem your having and not the reason to the
problem
Maybe the reason is something else
בתאריך 8 באוג׳ 2016 17:25, "Tammy Firefly" כתב:
Hi All,
We have asterisk 11.23 running sip to vitelity and from there IAX trunks
split off to where they
attempting to view the original CallerId?
>
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb <isr...@gmail.com> wrote:
> > Hi
> > Is there any configuration change in asterisk 13.9.1 to show original
> > callerid on a transfer
&g
Hi
Is there any configuration change in asterisk 13.9.1 to show original
callerid on a transfer
In asterisk 11.21 it works as expected
Thanks
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Hi
Could please show us your dialplan as you have it now and the lines in the
log on that call it would probably help a lot more
בתאריך 4 באוג׳ 2016 5:00 PM, "Nabeel" כתב:
> What happens when you dial "*98" from your own
>> phone.
>
>
> I get password prompt if a
Another nice sip packet is sngrep
Shows realtime the sip flows
But i think you have to chk the asterisk answer in the dialplan logic to
chk what context its hitting etc.
בתאריך 6 ביולי 2016 10:05 PM, "Steve Edwards"
כתב:
> On Wed, 6 Jul 2016, Victor Villarreal wrote:
Another thing i would check is encryption is disabled on the snom
בתאריך 8 ביוני 2016 10:07, "Israel Gottlieb" <isr...@gmail.com> כתב:
> Are you using stun? I have seen that when using stun
> בתאריך 8 ביוני 2016 09:54, "Faheem Muhammad" <faheem2...@gmai
Are you using stun? I have seen that when using stun
בתאריך 8 ביוני 2016 09:54, "Faheem Muhammad" כתב:
>
>
> Are you sure *nslookup *command is returning as expected?
> Also check the output of the below command.
> >> hostname && hostname -s && hostname -f
>
>
> On Tue,
e1} -${calltime}])
> exten=_X.,n,NoOp(diff)
>
> -
>
> Regards,
> Muhammad
>
>
> On Wed, May 18, 2016 at 5:05 PM, Israel Gottlieb <isr...@gmail.com> wrote:
>
>> Hi all
>>
>> Is there anywa
Hi all
Is there anyway i could get in the dialplan the amount of time a caller
waited in the queue before exiting?
Thanks
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alik <i...@pack-net.co.uk> wrote:
>
>>
>>
>> On 11 May 2016 at 10:24, Israel Gottlieb <isr...@gmail.com> wrote:
>>
>>>
>>> Hi all
>>>
>>> How is avg hold time and avg talktime calculated and over long a period
>>>
Hi allHow is avg hold time and avg talktime calculated and over long a period of time?Thanks,Israel
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if i remember correctly nerdvittles has a article
"google speech recognition api asterisk" brings results
On Tue, Feb 23, 2016 at 11:56 PM, Frank wrote:
> On Tue, 2016-02-23 at 17:06 +, Steve Howes wrote:
>
> > Google?...
>
> Yeah... searched "google voice
Try putting progress instead of answer
הודעה מקורית
מאת: Tony Mountifield
נשלח: יום רביעי, 25 בנובמבר 2015 08:14
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: [asterisk-users] Dialing a call back out on same SIP trunk as it came in
Use redirect
Using pjsip you can have multiple endpoints for each extension
הודעה מקורית
מאת: A J Stiles
נשלח: יום רביעי, 2 בספטמבר 2015 13:10
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] Single SIP User on multiple location
You could use the group functionCreate the group by extension and check how many calls are in the groupIf it's more than you allow then have it send a email
Look at the group function
It looks like you are dialing a external # then that won't work
הודעה מקורית
מאת: Luca Bertoncello
נשלח: יום שישי, 5 ביוני 2015 19:02
אל: asterisk-users@lists.digium.com
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] תשובה: Missed call
Israel
from more than one phone
Zitat von Israel Gottlieb isr...@gmail.com:
Shalom, Israel!
Using chan_sip you need to create another user aand then dial both
Using pjsip you can connect 2 devices
Thank you. Unfortunately it seems that I don't have pjsip available as
package on the OpenWRT where I
At the end of the Command you could use options one of them is the c (not
apital) which sends a cancel event to the phone
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Fri, Jun 5, 2015 at 9:53 AM, Luca Bertoncello lucab...@lucabert.de
wrote:
Zitat von Israel Gottlieb isr...@gmail.com
If you the c option in the dial command it will send answered else where sip message to the phone and most ip phones understand thatThe cell will always display a missed call
Using chan_sip you need to create another user aand then dial both
Using pjsip you can connect 2 devices
הודעה מקורית
מאת: Luca Bertoncello
נשלח: יום שישי, 5 ביוני 2015 09:24
אל: ML, Asterisk users
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: [asterisk-users]
We could probably parse the rdnis field to see if it that hop is on the list
הודעה מקורית
מאת: jg
נשלח: יום חמישי, 28 במאי 2015 12:18
אל: Asterisk Users Mailing List - Non-Commercial Discussion
השב ל: Asterisk Users Mailing List - Non-Commercial Discussion
נושא: Re: [asterisk-users] Seeking advice about ISDN BRI Cards
Thank you all for valuable input,
another
thanks for the reply
On Wed, Nov 12, 2014 at 8:03 PM, Matthew Jordan mjor...@digium.com wrote:
On Tue, Nov 11, 2014 at 1:43 PM, Israel Gottlieb isr...@gmail.com wrote:
well it should but this morning my database hosted at a remote location
was
down due to conditions at the remote site
Hi all
Does anyone know of a variable that i could check to see if the reason func
odbc didnt return results was because of a timeout error so i could play a
audio file about that
thanks
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On Tuesday, November 11, 2014, jg webaccounts...@jgoettgens.de wrote:
Why are you concerned? ODBC reconnects automatically if necessary.
jg
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right but that is the problem and i was wondering if there is way for
asterisk to set a variable when that happens just like curl
On Tue, Nov 11, 2014 at 5:37 PM, jg webaccounts...@jgoettgens.de wrote:
Unless of course the database server is not running at all for some reason.
But that's not
well it should but this morning my database hosted at a remote location was
down due to conditions at the remote site
the question isnt if it should happen or not
the questions is there a way for me to know that the odbc query retruned
empty because of a connection timeout?
in curl i could get
if you use a papt2 or so spa2101 then you could have alert info set to
different lengths or styles of ringers
i use that in a dorm with phones and have the phones ring short rings at
night so it wont wake up the students
On Tue, Aug 5, 2014 at 10:24 PM, Kevin Larsen
you could save the info in astdb for the last call per extension and then
pull it from there
On Tue, Jun 3, 2014 at 12:31 PM, Stefan Gofferje li...@home.gofferje.net
wrote:
Hi,
I would like to implement an auto-redial function in a context. The idea
is about like this:
Dial a number
Hear
Hi all
asterisk 1.8.23
I have odbc all setup to mysql but cant figure out why the dialplan wont
write to the odbc function
fubc_odbc.conf
[DEVICES]
dsn=device-conn;dsn in res_odbc not odbc.ini
readsql=SELECT call.callNum, call.city, devices.callId, devices.id FROM
call INNER
http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback
On Fri, Jun 1, 2012 at 1:45 PM, Satish Barot satish4aster...@gmail.comwrote:
I believe you want your caller to request for a callback while he/she
waits in a queue and when your agents are free, you want to call him back
and place
are you by chance using the a2billing script?
On Mon, Apr 2, 2012 at 5:43 PM, Syco syco...@gmail.com wrote:
No, I don't do transcoding, I've disabled all the codec except for the
g729.
But in my last test I've found out what is the problem (not yet how to
solve it)
I make all my calls
that bug is running since the start of 1.8 and has been fixed in 1.8.9
https://issues.asterisk.org/jira/browse/ASTERISK-17474
i know it says that after the first time asterisks starts it works but
thats true only if the moh was loaded before the timing
its a long story but the fix is finally in
wow i just tried in hebrew and i'll say just 1 word WOW
On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote:
On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:
Works beautifully. Amazing job Lefteris. Thanks.
The best result I got in probability was 0.9725632 by saying, hello.
On Mon, Dec 26, 2011 at 9:03 AM, Steve Edwards asterisk@sedwards.comwrote:
On Mon, 26 Dec 2011, isr...@gmail.com wrote:
Rename the wav to ulaw
Miss_audio.ulaw
Very bad advice.
that might be but if you take a pcm ulaw encoded file and name it .wav
asterisk will throw that error
I
On Wed, Sep 14, 2011 at 5:27 AM, Dale Noll dn...@wi.rr.com wrote:
On 09/13/2011 07:49 PM, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/**custom/${TOPMENU})})
im trying to see if a file is available
On Wed, Sep 14, 2011 at 4:08 AM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 14 Sep 2011, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/**
custom/${TOPMENU})})
im trying to see if a file
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using agi
asterisk 1.6.2.20
thanks
--
On Thu, Sep 8, 2011 at 1:52 AM, Israel Gottlieb isr...@gmail.com wrote:
On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote:
Hi all
i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the
returned
Hi all
i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the returned
info is English or digits then the callerid name field gets populated with
that but if the returned info is utf8 like Hebrew then the callerid field
remains
On Thu, Sep 8, 2011 at 1:47 AM, Israel Gottlieb isr...@gmail.com wrote:
Hi all
i have a very weird problem with curl and utf8 characters
i'm trying to do a cnam lookup from a web-service with curl if the returned
info is English or digits then the callerid name field gets populated
Set(VOLUME(TX)=10) is correct but you arent putting it in a context so
asterisk doesnt know how to deal with it
do this
[bigbluebutton]
exten = _.,1,Set(VOLUME(TX)=10)
exten = _.,1,Set(VOLUME(RX)=10)
exten = _.,n,Goto(start-dialplan,s,1)
exten = _.,n,Hangup
On Thu, Aug 4, 2011 at 4:33 PM,
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote:
On Mon, Jul 18, 2011 at 07:58, Steve Daviesdavies...@gmail.com wrote:
The magic sauce that you need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your
On Fri, Jul 22, 2011 at 12:50 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 07/21/2011 04:43 PM, Israel Gottlieb wrote:
On Fri, Jul 22, 2011 at 12:39 AM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
On 07/21/2011 04:34 PM, Joaquin Sosa wrote
user-agent could be set in sip.conf
On Wed, Jul 20, 2011 at 12:43 PM, Alex Balashov
abalas...@evaristesys.comwrote:
On 07/20/2011 05:00 AM, Masood Ahmed wrote:
Hello All, Is there any one who can help me to change the From
field parameters in option packets, I have seen that in option
Hi all
i have a scenario where i have 2 DSL lines (i know its not that reliable but
it fits the bill) connected to 1 box and would like my isp to round robin
between both dsl (to allow for more capacity - each dsl could get me thru
about 16-18 calls and i need about 30
incoming sip gets routed
Hi
Does anybody have a idea how I could set sip headers when using call files?
I have to call out a specific trunk so I cant use local as the trunk
what i'm trying todo is send out calls as anonymous but at the itsp it
should be filed as being called out thru a specific DID and not the main DID
On Thu, Apr 14, 2011 at 3:51 PM, Israel Gottlieb isr...@gmail.com wrote:
Hi
Does anybody have a idea how I could set sip headers when using call files?
I have to call out a specific trunk so I cant use local as the trunk
what i'm trying todo is send out calls as anonymous but at the itsp
How could i check if the call is using t38 except looking at the sip debug?
Is there any variable thats set or could be set?
thanks
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t38 ?
Thanks
On Mon, Mar 28, 2011 at 1:10 AM, Larry Moore lmo...@starwon.com.au wrote:
On 28/03/2011 5:48 AM, Israel Gottlieb wrote:
still no luck
i hear it change to t38 but it just doesnt connect
Do you have two fax devices at your end, even a fax-modem attached to a
computer
still no luck
i hear it change to t38 but it just doesnt connect
On Sun, Mar 27, 2011 at 5:26 AM, Larry Moore lmo...@starwon.com.au wrote:
Perhaps this will help.
I have a SPA8800 which has 4 x FXS 4 x FXO ports.
It took me some time to produce a working configuration.
In Asterisk I
Hi
I'm trying to get the spa 8000 used with a fax machine using t38 passthru
i have tried with 1.6.2 and 1.8.3 and is still a no go
the provider i use is 012 in israel wich supports t38 (i use it with ffa)
could anybody give me a clue how to get this working if it should
t38pt is set to yes in
seconds.
On Wed, Feb 23, 2011 at 3:17 PM, Israel Gottlieb isr...@gmail.com wrote:
Hi
Does anyone know how i could extend the timer for the ringing time on a
pri or sip trunk ?
Today the call gets a cancel request after a minute if not answerd yet
is it on asterisk or is a provider side
Hi
Does anyone know how i could extend the timer for the ringing time on a pri
or sip trunk ?
Today the call gets a cancel request after a minute if not answerd yet
is it on asterisk or is a provider side setting?
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but I cant
seem to find it).
Thanks,
Israel Gottlieb
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