Hi
Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :
[...]
- Even if direct_media is disabled: Is there a way to make Asterisk
always use a common codec between SIP endpoints,
so it doesn't need to transcode?
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
--
Le 01/10/2019 à 16:38, Eric Wieling a écrit :
Verify ${myCpt} is not empty.
Yes, it was that. Many thanks
On 10/1/19 10:24 AM, Administrator TOOTAI wrote:
Hi list,
on asterisk 13 I use
same => n,Set(__myCpt=$[${myCpt} + 1])
which is working well. On an Asterisk 16 I get, for this s
Hi list,
on asterisk 13 I use
same => n,Set(__myCpt=$[${myCpt} + 1])
which is working well. On an Asterisk 16 I get, for this same command
[2019-10-01 16:15:01] WARNING[28197][C-0008]: ast_expr2.fl:470
ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '+',
expecting
Le 30/09/2019 à 15:58, Joshua C. Colp a écrit :
On Mon, Sep 30, 2019, at 10:52 AM, Administrator TOOTAI wrote:
Le 30/09/2019 à 11:45, Joshua C. Colp a écrit :
On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote:
Hi list,
I would like to now what is the sense of such type of entry
Le 30/09/2019 à 11:45, Joshua C. Colp a écrit :
On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote:
Hi list,
I would like to now what is the sense of such type of entry in security.log
[2019-09-27 15:12:24] SECURITY[26964] res_security_log.c:
SecurityEvent="ChallengeSent&quo
Hi list,
I would like to now what is the sense of such type of entry in security.log
[2019-09-27 15:12:24] SECURITY[26964] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic
e="PJSIP",EventVersion="1",AccountID="",
Le 13/09/2019 à 14:03, Brian J. Murrell a écrit :
How can I use an IF statement with a true value being a variable that
has a colon in it? The colon in the true value variable is being taken
as the delimiter for the false value.
The only solution I came up with was some hackery to use
figuration_res_pjsip-endpoint_direct_media_method
BR
Jöran
On Thu, Aug 15, 2019 at 2:03 PM Administrator TOOTAI <mailto:ad...@tootai.net>> wrote:
Le 15/08/2019 à 13:22, Jöran Vinzens a écrit :
> Hi All,
>
> We are using asterisk 16.5 and having an issue with the first
Le 15/08/2019 à 13:22, Jöran Vinzens a écrit :
Hi All,
We are using asterisk 16.5 and having an issue with the first re-invite
after the call has been established.
We can see the call gets up and you see in the logs the bridge type has
changed and after that a re-invite is triggered.
Is
Hello list,
I'm looking for a solution that can be applied to a stock asterisk 16
(pjsip if it matter) running Debian 9 (php7.0).
Statistics should be available for normal calls and queues using a WEB
interface. Open source better but not necessary,
Any feedback appreciate, no matter if
Le 10/06/2019 à 10:53, Benoit Panizzon a écrit :
What about to put eveything in a variable and the remove the last
character if it equal &
Yes, I considered this...
What if you dial three endpoints and the middle one (or last one) is
empty? You would also need to remove the first & and any
Le 09/06/2019 à 13:19, Benoit Panizzon a écrit :
Dear List
Hello
It's probably been more than a year now I switched from chan_sip to
pjsip. pjsip works much cleaner than chan_sip.
But!
I have come across a Problem I was not able to solve with Asterisk
Dialplan Logic.
With pjsip an
Le 08/06/2019 à 05:20, John T. Bittner a écrit :
Hopefully, this helps someone else.
This seems to be working for me.
# Fail2Ban configuration file
[INCLUDES]
#before = common.conf
[Definition]
failregex = NOTICE.* .*: Request \'REGISTER\' from '.*' failed for
':.*' .* - No matching
Le 08/04/2019 à 20:06, Administrator TOOTAI a écrit :
Hi,
I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk
13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also
with PJSIP. Both LAN Asteriks are also connected via IAX.
Everything is working fine
Hi,
I have following setup: Asterisk 1.4 (IP 10.1.1.250) connect to Asterisk
13 (IP 10.1.1.251) with PJSIP, this one connected to the provider also
with PJSIP. Both LAN Asteriks are also connected via IAX.
Everything is working fine except SIP call from 1.4 to external number:
there is no
Hi all,
I switched an old asterisk 1.8 to a new 13 version, stock version from
Ubuntu 18.04 server.
I did some modification in dialplan but after a reload they are not
taken in account :(, even after restarting asterisk.
I checked logs and found lots of
merging incls/swits/igpats from
Le 15/03/2019 à 15:18, sean darcy a écrit :
From my provider I get extensions of:
+1<10digit number>
1<10 digit number>
<10 digit number>
seemingly randomly.
What I'd like to do is
exten=_!1234567890,1,Answer()
which would match anything ending in 1234567890.
But that doesn't work since !
Hello,
since ipv6 doesn't really push ipv4 out of our networks, we still use
IAX (when possible) to not face NAT problems from SIP, Asterisk 16 included.
Question is, what is the future of IAX. Is it a dead protocol ? Will it
stay as is even only security problems resolved ? Will be (sooner
Le 11/03/2019 à 10:23, Marcelo Terres a écrit :
Hello Jean-Denis.
I believe the idea is that you answer the survey for each type of
scenarios you are running.
So one for call centre, another one for ivr, etc...
And what for instance about exact version of asterisk?
We are in the same
Hi,
I rey to register an Asterisk 16.2.1 pjsip to an ASTERISK 13.25.0
chan_sip using ipv6 and pjsip_wizard.
I only got it work if in remote_hosts I put the ipv6 address and not the
hostname like sip.domain.ltd No need to say that an entry is
existing for the hostname in DNS.
BTW, how
Hi,
We upgraded an Asterisk 11 server to 16.1.1, going from chan_sip to
pjsip, on a site using Gigaset phones. They are registring well despite
the fact that we get a lot of errors like
[Feb 22 18:30:07] ERROR[1556]: pjproject: : sip_transport. Error
processing 367 bytes packet from UDP
Le 14/02/2019 à 00:12, sean darcy a écrit :
I'm using BLACKLIST() to check numbers, which does not like leading +
signs. I want to test if there is a plus sign, and then remove it.
I tried:
; strip leading plus sign
same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
same =>
Le 01/02/2019 à 09:59, basti a écrit :
Hello,
Hi
my Asterisk is installed on my router. From my ISP I only get an dynamic IP.
In sip.conf I have try:
externhost=host1.mydns.unix-solution.de
externrefresh=300
but after reconnect I cant call from "outside".
asterisk*CLI> sip show registry
Le 30/01/2019 à 05:17, Jose Tavares a écrit :
Hi guys ..
I have some experience with asterisk and sip since I have been using it
for over 10 years.
But in the last years I have been just maintaining the installations we
have without updating myself on the new features of it.
Now I have a
Le 17/01/2019 à 10:38, Olivier a écrit :
Hello,
Hi
These questions crossed my mind this morning :
In general, are anonymous international calls allowed (ie calling from
one country to a number in an other country while hiding your own caller
id) ?
Are there special rules in Europe for
Hello,
I have an external agent which register dynamically in a queue and I
setup his PJSIP account on a identity of a local phone which is
configured to redirect all calls to the agent. Agent can have only on
call at a time.
When agent takes a call he is paused from the queue to avoid
Le 11/01/2019 à 10:23, Neil Youngman a écrit :
On 11/01/2019 09:19, Administrator TOOTAI wrote:
Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by
a fax machine. It was suggested that having a "fax" extension and
&quo
Le 11/01/2019 à 10:12, Neil Youngman a écrit :
A while back, I posted about detecting when a call was picked up by a
fax machine. It was suggested that having a "fax" extension and
"faxdetect=yes" would cause it to jump to the "fax" extension. This was
not something I could get to work.
I
Le 10/01/2019 à 16:18, Social Boh a écrit :
Hello,
maybe:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_DEVICE_STATE
Thanks Social, exactly what I wanted.
---
I'm SoCIaL, MayBe
El 10/01/2019 a las 09:13, Administrator TOOTAI escribió:
Hi,
on an Asterisk 16
Hi,
on an Asterisk 16 with PJSIP I want to know the state of a device (idle,
busy, unavailable, ...) in the dialplan. I tried with ChanIsAvail() but
this one doesn't return the real state (eg a device calling an extension
which is running ChanIsAvail() is marked as idle!)
When I use in a
Le 27/12/2018 à 20:42, Social Boh a écrit :
Hello,
you have to use GROUP and GROUP_COUNT functions.
Well, could be done for extensions but for queue ? Does it mean
ringinuse is useless ?
[...]
El 27/12/2018 a las 14:14, Administrator TOOTAI escribió:
Hello,
I'm used to set call-limit
Hello,
I'm used to set call-limit in sip.conf Now I switched one customer
Asterisk to 16 version and can't get the behavior back, as well for
extensions as for queues.
I set ringinuse=no for queues and have max_audio_streams = 1
max_video_streams = 0. I wanted to add max_calls = 1 but this
Le 07/12/2018 à 15:56, hw a écrit :
On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:
Le 07/12/2018 à 14:32, hw a écrit :
[...]
Queues seem to be the only way to have several phones ring at once,
or are there other ways?
Dial(SIP/Phone1/Phone2&.../Phonex,,)
Good to know, th
Le 07/12/2018 à 14:32, hw a écrit :
[...]
Queues seem to be the only way to have several phones ring at once, or
are there other ways?
Dial(SIP/Phone1/Phone2&.../Phonex,,)
--
Daniel
--
_
-- Bandwidth and Colocation
Le 27/11/2018 à 13:18, Joshua C. Colp a écrit :
On Tue, Nov 27, 2018, at 8:13 AM, Administrator TOOTAI wrote:
Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :
On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]
[TOOTAiAudio]
;
; Call our gateway
exten = s,1,Set(PJSIP_HEADER
Le 27/11/2018 à 12:13, Joshua C. Colp a écrit :
On Tue, Nov 27, 2018, at 5:49 AM, Administrator TOOTAI wrote:[...]
[TOOTAiAudio]
;
; Call our gateway
exten = s,1,Set(PJSIP_HEADER(add,X-TOOTAiAudio-CALLED)=${ARG1})
same = n,Dial(PJSIP/${ARG1}@TOOTAiAudio,,T)
same = n,Return
exten = h,1
Hi list,
to manage an external queue agent the only solution I found is to
connect a local account and redirect calls to this account using forward
features from the phone (SNOM). The problem I face is that before
calling the agent I would like to set extra header. Dialplan to call
external
No one on this ?
Le 22/11/2018 à 17:59, Administrator TOOTAI a écrit :
Hi all,
I want to set dynamic queue with non local members. I create an
extension 115 in [localEP] context which is doing the job, eg calls to
this extension are forwarded to the non local endpoint (which is an IP
phone
Hi all,
I want to set dynamic queue with non local members. I create an
extension 115 in [localEP] context which is doing the job, eg calls to
this extension are forwarded to the non local endpoint (which is an IP
phone connected to an external Asterisk 13 version). Phones are SNOM.
Queue
Le 20/11/2018 à 19:50, Administrator TOOTAI a écrit :
Hi,
I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a
problem with user variable defined in sip.conf using setvar. It work
like a charm -even on asterisk 13 version- but can't get it work in 16.
The variables are defined
Hi,
I'm on the way to upgrade a dialplan from 1.8 to 16.0.1 and face a
problem with user variable defined in sip.conf using setvar. It work
like a charm -even on asterisk 13 version- but can't get it work in 16.
The variables are defined in pjsip with set_var and a pjsip show
endpoint does
Hi list,
I have a queue in which I add a member located outside the company and
connected to an outside asterisk. Let's say peername is ABCD123. In the
queue I gave SIP/ABCD123 as interface which is not existing on the local
asterisk.
Is there a way to connect a member from a queue which is
Le 27/07/2018 à 09:36, Stefan Viljoen a écrit :
Hi all
This is a followup on my post "Asterisk 13 - system() dialplan app cannot call bash
scripts" from yesterday
I've given up trying to use system() to call BASH scripts with parameters from
Asterisk 13.
Turned out under Asterisk 13.22.0
Hello,
we have 4 asteriks, 2 in office on one server (wazo and mobydick), and 2
in DC (self compiled) each on his own server. All of them are VMs under
Debian Stretch. We used OpenVPN to connect the machines together in TAP
mode, everything was running well.
Setup is following: the 2
Hi all,
is it possible to access PJSIP configuration variables from the dialplan
? Exemple: I want to get the username of a type = auth context.
Thanks for any hint
Daniel
--
_
-- Bandwidth and Colocation Provided by
Le 16/04/2018 à 16:52, Joshua Colp a écrit :
On Mon, Apr 16, 2018, at 11:47 AM, Administrator TOOTAI wrote:
Hi all,
we are trying to move our servers from chan_sip to chan_pjsip. At this
time no problems with phones, they all register fine and can place
calls. But for a trunk we face problem
Hi all,
we are trying to move our servers from chan_sip to chan_pjsip. At this
time no problems with phones, they all register fine and can place
calls. But for a trunk we face problem and can't place calls despite the
fact that registration is OK. What we get is:
[2018-04-16 16:08:33]
Hi Herrmann
Le 12/04/2018 à 17:22, Hermann Wecke a écrit :
I'm trying to solve a mystery for the last couple of days.
I have a mix of D70, D50 and D40 behind NAT. Server is in a
colocation, not a VPS.
For several years, everything was working fine, no issues. A few days
ago I started having
Le 26/03/2018 à 11:08, Antony Stone a écrit :
On Monday 26 March 2018 at 10:58:01, Administrator TOOTAI wrote:
Hi all,
we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in
datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances
behind FW. Problem we face
Hi all,
we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in
datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances
behind FW. Problem we face is that when we reboot the DC Asterisks, the
trunks (SIP or IAX) become alive from DC Asterisks to clients ones
Le 25/01/2018 à 10:37, Atux Atux a écrit :
Being honest, i did not manage to make it work. Now whoever calls the
system extensions, does not know if they are on another phone call or
away from the office.
For chan_sip you can do like this before ringing an extension. Status is
returned in
Hello list,
I want to connect 2 sites both having asterisk installed (1.4 and
13.16from Ubuntu 14.04). When calling from 13.16 to 1.4 (call to echo
test which should show video) I get in logs
[2017-06-13 14:45:26] WARNING[17176][C-03b0] channel.c: Unable to
find a codec translation
Le 08/06/2017 à 15:15, J Montoya or A J Stiles a écrit :
On Thursday 08 Jun 2017, Olivier wrote:
Hello,
I'm building a new Asterisk system from source on Debian Stretch.
My building script fails as package libmyodbc is currently missing from
Debian Stretch repo.
Is there a work around this
Le 06/06/2017 à 16:25, Daniel Tryba a écrit :
On Tue, Jun 06, 2017 at 03:18:32PM +0200, andre castro wrote:
extensions.conf:
[home]
exten = 102,1,Answer()
same = n,Wait(1)
If this is copy and paste, then your dialplan is broken (= should be =>)
Well, no. = or => are the same.
--
Daniel
Le 15/05/2017 à 17:34, Tech Support a écrit :
All;
I have an application that dials a list of numbers and then plays a
recorded message. My customer uses it to dial a list of customers to
confirm their appointment for the next day. No biggie, maybe 25 – 30
calls per day for customers who
Le 23/03/2017 à 20:17, Jonas Kellens a écrit :
Hello
is there any more information on how to reload/read musiconhold files ?
CLI> module reload res_musiconhold
--
Daniel
On 07-03-17 10:46, Jonas Kellens wrote:
Hello
I did not mention it but of course the MOH directory is listed in
Hi all,
we have a running Asterisk 11.25.1 in a VM (qemu/kvm) OS being Debian
7.11 (wheezy), the host OS being the same.
Problem: when we restart the server (eg host + VM), all customers
Asterisk connecting without a VPN (doesn't matter which Asterisk
version) are no more reachable. Same
Le 26/10/2016 à 12:21, Joshua Colp a écrit :
Administrator TOOTAI wrote:
Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk
11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub
Le 25/10/2016 à 23:10, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk 11.24.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.24.0 resolves several
=> 201,n,NoOP(My variable is ${myvar})
Thanks for the information. As on voip-info.org this function is no more
detailed and status is 1.2 I thought that it was removed from latest
version.
On Mon, Oct 24, 2016 at 9:33 AM, Administrator TOOTAI <ad...@tootai.net
<mailto:ad...@tootai.ne
Hi list,
is there any existing IAX command to add information to a call like
SipAddHeader? Another solution is sending text frame (0x07) frame type,
but I don know how do it in a dialplan.
Thanks for any hint.
--
Daniel
--
Le 15/10/2016 à 18:17, Jerry Geis a écrit :
I have a host 192.168.1.3 that wants to run SIP on 5068 (long story).
My host is 192.168.10.201.
My host needs to stay on 5060 because of all the other devices I have
connected.
I tried putting port=5068 in my SIP extension definition but that did
not
Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :
Hi,
If you're not using RTP encryption did you uncheck the option in your
RTP TAB from identity ?
This is the log. ex dialling 0 from snom phone
<--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878
Le 02/09/2016 à 11:26, Jonas Kellens a écrit :
Hello
when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :
[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now
Le 01/09/2016 à 17:27, D'Arcy J.M. Cain a écrit :
On Thu, 1 Sep 2016 13:49:57 + (UTC)
t...@softins.co.uk (Tony Mountifield) wrote:
What module am I missing?
The ExecIf command is provided in the module app_exec, which is
usually located at /usr/lib/asterisk/modules/app_exec.so
Yes, I
Le 01/09/2016 à 03:57, D'Arcy J.M. Cain a écrit :
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <ad...@tootai.net> wrote:
Something like
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}&
Le 30/08/2016 à 18:05, D'Arcy J.M. Cain a écrit :
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <ad...@tootai.net> wrote:
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NO
Le 30/08/2016 à 15:56, D'Arcy J.M. Cain a écrit :
I have an extension that looks like this:
exten => 55,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1/user2/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say,
Le 26/04/2016 17:23, Mamadou NGOM a écrit :
Hello,
Having installed DAHDI to be able to use the meetme() application , when
I start the dahdi service it generates me the following error:
-bash: /etc/init.d/dahdi: No such file or directory
Clear, the file dahdi is not existing. Did you copy
Le 18/03/2016 16:20, Trey Hilyard a écrit :
I am trying to set up my Asterisk server so that it will recognize an
incoming call to the Asterisk's own Location Routing Number (LRN),
validating the "rn" in the INVITE and then using the Called Number from
the INVITE as the extension in the
Le 15/03/2016 11:20, Feroz Ahmed a écrit :
Hi I need help
Hello Ahmed
This is the error:
[...]
[Mar 14 19:55:15] WARNING[20595][C-000b]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
== Everyone is busy/congested at this
Le 10/03/2016 15:12, Karl Anderson a écrit :
Is is possible to use variables in followme.conf? Is it possible to
include another conf file?
in the followme.conf file
#include local/followme.d/*.conf
will include all files having .conf extension in this directory
--
Daniel
--
Hi Vitor
Le 12/02/2016 19:17, Vitor Mazuco a écrit :
I think that my monden is locked for Voice
I use a Huawei E173, someone know how can I unlock it?
Is necessary to upgrade the firmware?
Google is your friend, plenty of informations like
Hello,
We installed the subject detailed versions on a uptodate debian wheezy.
When starting Asterisk we get
Loading chan_pjsip.so.
== Registered RTP glue 'PJSIP'
== Registered channel type 'PJSIP' (PJSIP Channel Driver)
18:26:10.812 sip_endpoint.c !Module "mod-refer" registered
Le 17/09/2015 12:37, Дорофеев Сергей a écrit :
Hello list!
Hello
Sorry for kinda dumb question, I guess, but I have too little time to
research it by myself.
I have a SIP packet, which looks like this:
<--- SIP read from UDP:10.186.0.38:5060 --->
INVITE
Le 07/08/2015 23:54, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk 11.19.0.
[...]
Hello,
We have problem with patches since 11.18.0 We have to download the full
tar.gz to get last version :-(.
Before this, since ages, we used to
Le 09/07/2015 17:05, Tzafrir Cohen a écrit :
On Thu, Jul 09, 2015 at 12:28:15AM +0200, Administrator TOOTAI wrote:
zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1
../asterisk-11.18.0-patch
patching file .version
Hunk #1 FAILED at 1.
1 out of 1 hunk FAILED -- saving rejects to file
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual
running version. Patch failed with
zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0
../asterisk-11.18.0-patch
can't find file to patch at input line 5
Perhaps you used the wrong -p or --strip option?
The
Le 08/07/2015 17:36, Richard Mudgett a écrit :
On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net
mailto:ad...@tootai.net wrote:
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0
actual running version. Patch failed with
zone-s
Hi,
we think that there is a mistake with the asterisk-11.18.0.patch. The
file look like
diff --git a/.version b/.version
index c5df2aa..150754a 100644
--- a/.version
+++ b/.version
@@ -1 +1 @@
-11.18.0-rc1
\ No newline at end of file
+11.18.0
\ No newline at end of file
[...]
trying to
Le 01/05/2015 00:05, Andrew Martin a écrit :
- Original Message -
From: Administrator TOOTAI ad...@tootai.net
To: asterisk-users@lists.digium.com
Sent: Thursday, April 30, 2015 4:43:33 PM
Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In
I am running Asterisk
Le 30/04/2015 19:18, Andrew Martin a écrit :
Hello,
Hello
I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and internal
phones are located on the 10.10.32.0/21 LAN subnet. I have many internal SIP
phones, which appear to be working correctly. I have a few external phones
Hello,
I setup a door open system with a basic DTMF card. The card is connected
to an Sipura/Linksys 3102 FXS port and is powered by this port.
My problem is that when I send a call with Dial() command, channel has
to be answered before receiving DTMFs, what my card does not. Is there a
way
Hello,
on previous versions of asterisk, extension h and H make us know who
ended a call (caller or callee). In the last * versions, seems that only
h extension is used, as stated here
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
In the last versions, how do we know which
Hi,
Le 12/03/2015 17:28, Salaheddine Elharit a écrit :
hello list,
i use the code below
[macro-chanspy]
exten = s,1,Authenticate(${ARG1})
exten = s,n,ChanSpy(SIP/${EXTEN:3},__dqs)
Here you have a problem: ${EXTEN} value is s
[...]
Daniel
--
]
exten = fax,1,Dial(IAX2/300,,)
same = n,Hangup
exten = _X.,1,Wait(10)
same = n,Congestion()
Not particulary clean.
Daniel
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent
of the path.
Asterisk version is 11.15.0 from Elastix. Same happend on a stock 11.16.0
Thanks for your answer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Wednesday, February 18, 2015
Hello
Le 17/02/2015 17:00, Administrator TOOTAI a écrit :
Hi,
as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.
It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.
On the upper Asterisk gw it's
Hi,
as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.
It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.
On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the
Hi,
we have following setup:
fax machines PSTN - GW SIP - Asterisk - Peer IAX (Elastix)
Asterisk is 11.16 as well as Asterisk version of Elastix peer.
When sending incoming fax calls to the Peer IAX -which receive them
using hylafax- we want to tell our asterisk to NOT detect fax CNG. It's
Le 28/01/2015 22:03, Steven McCann a écrit :
Hello,
Hi
I'm investigating a situation where there was a hundreds of minutes of
calls from an internal SIP extension to an 855 number in Cambodia,
resulting in a crazy ($25,000+) bill from the phone company. I'm
investigating, but can anyone
and there is no audio leg yet).
It works perfectly :-), many thanks for the tip.
Regards
--
Daniel
[...]
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Friday, January 16, 2015 5
Hello,
our gateway receive incoming calls from an outside gateway for multiple
DIDs. For some of them we want fax detection, for other no. To do so,
faxdetect is set to yes, but how to disable the fax detection for a
specific incoming DID? For those DIDs, we just want to forward the call
to
Hi all,
On an Elastix server with asterisk 11.13.0 I have no verbose logs
despite the fact that it's OK in CLI, eg verbose set to 3 in my case
Logger.conf
[logfiles]
;
; Format is filename and then levels of debugging to be included:
;debug
;notice
;warning
;error
;
Le 01/10/2014 11:40, Olivier a écrit :
Hi,
Hi
Someone reported me that from a PBX on which someone gained fraudulent
access, he could observe hundreds of calls to the same destination
number.
For curiosity's sake, I'm wondering why would this happen (dialing the
same number over and over)
Le 13/09/2014 20:04, Alok Srivastava a écrit :
*Dear List*
Plz help, i am not much experienced with asterisk. i configured it on
ubuntu 12.04. no problem when i call any mobile no(0091XX) but
when i call on my local asterisk no.(101,102 or 105) it is not
connecting giving error
Auto
Le 02/09/2014 08:47, Nick Awesome a écrit :
Hello guys.
Hi
Have 2 external numbers that required registration on provider server,
trunk1: 734322600*05*@80.75.132.66
trunk2: 734322600*50*@80.75.132.66
Thing is I can’t figure out how to route them to different IVRs
by default Asterisk
Le 02/09/2014 09:38, Nick Awesome a écrit :
So there is no way to do that with pjsip?
Sorry, I didn't read carefully the subject. I can't answer for pjsip. My
bad :-(
On 02 Sep 2014, at 11:35, Administrator TOOTAI ad...@tootai.net wrote:
Le 02/09/2014 08:47, Nick Awesome a écrit
Le 02/09/2014 20:18, Khalid Touati a écrit :
so it seems Asterisk Versions does not support video I guess
Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 with
GrandStream phones (H263, H263+ and H264). Works perfectly
On Mon, Sep 1, 2014 at 9:30 AM, Khalid Touati
Hi,
is there a way to put a conference participant in talk only mode (not
listening) using CLI or AMI like mute/unmute ?
MeetMe in Asterisk 1.8
Thanks for any hint.
--
Daniel
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