On 8/15/16 3:16 PM, Jonas Kellens wrote:
Hello
after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in
MySQL DB) register anymore.
[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
I am having a problem with Fanvil phones (X3) when they make a call
through DAHDI. Pure SIP calls flow normally but when a call goes
through a DANDHI interface to the PSTN we only get one way audio. This
is Asterisk 13.10.0 (bundled pjsip) and Dahdi 2.11.1 with an Openvox
A400 card (4
On 8/15/16 11:04 AM, Eric Wieling wrote:
"make config" should also install the init script.
On 08/15/2016 11:36 AM, Jerry Geis wrote:
>On my Fedora 24 system, the "dahdi-tools" package contains an old-style
>init script /etc/rc.d/init.d/dahdi, and this seems to work just
fine with
I keep getting messages like these in the cli:
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql:
Realtime table general@ps_contacts: column 'qualify_timeout' cannot be
type 'int(10)' (need char)
[Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1246 require_mysql:
Anyone know a good replacement for phpagi? Unfortunately
development stalled long ago and it has not been updated. What is the
best solution for AMI and AGI on PHP? Thanks.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
On 7/20/16 9:58 AM, Faheem Muhammad wrote:
Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the
dial command breaks and the call control go to hangup block instead of
next priority. The error in CLI says
Until Asterisk 11 I could use sip.conf to set defaults for all
phones (language, dtmf, vmexten, etc) and just leave many fields in the
database as NULL. What would be the proper way to do this for Asterisk
13 and PJSIP?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
On 7/12/16 9:27 PM, George Joseph wrote:
On Tue, Jul 12, 2016 at 11:55 AM, Carlos Chavez
<cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote:
I am still a little confused about how to activate MWI with
PJSIP on Asterisk 13.9.1. I use realtime for c
I am still a little confused about how to activate MWI with PJSIP
on Asterisk 13.9.1. I use realtime for configuration. So far I have
tried setting both the mailboxes field on ps_endpoints and the mailboxes
field in ps_aors but I cannot get the indicator lamp to blink on any of
my phones
On 2016-07-02 15:16, Derek Bolichowski wrote:
> Hi Leandro,
>
> I believe if you check /usr/local/src/astersisk-13.9.1/contrib/mysql you will
> find a .SQL file that would build the default tables for you.
>
> Looking in the file, it appears there is a table created called `sippeers`
>
On 6/6/16 10:55 AM, A J Stiles wrote:
On Monday 06 Jun 2016, Markus wrote:
Hi AJ,
Am 06.06.2016 um 10:14 schrieb A J Stiles:
But why not call an AGI script, have this check the caller ID against a
MySQL database and return a status -- blocked or not -- in a variable?
Then you can manage
I use realtime for my Asterisk configuration and are now making the
transition to Asterisk 13 and PJSIP. I used alchemy to set up my
databases and I can now configure my endpoints. While trying to
configure a trunk I can see that there is a database table called
ps_registrations that
Asterisk 13.9.1 seems to be ignoring my realtime IAX
configuration. I have carried this configuration over from version 1.8
and it worked until 13.7 at least. The config mapping is done:
pbxoficina*CLI> core show config mappings
Config Engine: mysql
===> ps_contacts (db=general,
On 2016-05-18 16:32, Neeraj Chand wrote:
Hi All,
Has anyone used hints in realtime ?
(As in storing and loading hints from odbc)
I cannot find a table structure for this anywhere...?
Thanks
Neeraj
Hints are defined in the dialplan so if you are loading your dialplan
from a database it
I am having a strange problem with Asterisk 13 on a CentOS 7
plataform. I have several servers running on this configuration but a
particular installation on a Dell PowerEdge 220 server is the one giving
me the most problems. All installations are automated via a script so
there is no
On 4/7/16 11:55 AM, jg wrote:
Since a couple versions back I keep getting these messages when
compiling Dahdi:
make[2]: Entering directory
`/usr/src/kernels/3.10.0-327.13.1.el7.x86_64'
INSTALL
/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_sec.ko
Can't
Since a couple versions back I keep getting these messages when
compiling Dahdi:
make[2]: Entering directory `/usr/src/kernels/3.10.0-327.13.1.el7.x86_64'
INSTALL
/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi.ko
Can't read private key
INSTALL
On 4/6/16 2:39 PM, Duncan Turnbull wrote:
On 7/04/2016, at 6:01 AM, Carlos Chavez <cur...@telecomabmex.com> wrote:
On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network
On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking
On 4/5/16 3:17 PM, Joshua Colp wrote:
Carlos Chavez wrote:
I am currently having a voice quality problem with one of our Asterisk
servers. We have checked the network and we have found no problems that
could cause the voice to sound cracked and with small interruptions. I
am looking
I am currently having a voice quality problem with one of our
Asterisk servers. We have checked the network and we have found no
problems that could cause the voice to sound cracked and with small
interruptions. I am looking at the timing source for Asterisk and it is
currently using
terisk?
On Thu, Mar 24, 2016 at 12:59 PM, Carlos Chavez
<cur...@telecomabmex.com>
wrote:
We've been having some problems with an E1 PRI line for a few days.
We
get the following errors:
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
[Mar 24 10:13:39]
We've been having some problems with an E1 PRI line for a few days. We
get the following errors:
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2 ROSE REJECT:
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c: PRI Span: 2INVOKE
ID: 316
[Mar 24 10:13:39] ERROR[22009] chan_dahdi.c:
On 2016-03-13 02:30, Recursive wrote:
On 07.03.2016 20:28, George Joseph wrote:
The current Asterisk 13 and master git branches have a new feature
that will be included in 13.8.0: The ability to compile and run
Asterisk with a bundled version of pjproject.
[...]
PLEASE TRY THIS!! I'd love
On 2016-03-13 07:01, Mehdi Shirazi wrote:
Hi
This is my system.conf :
dynamic=loc,1:0,31,0
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
dynamic=loc,1:1,31,0
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
and this is my chan_dahdi.conf:
group=0
echocancel = yes
I am having a problem trying to use the realtime database for
musiconhold for Asterisk 13. Everything is setup and I can see the mapping:
===> musiconhold (db=general, table=musiconhold)
Only what is in the musiconhold.conf file appears in Asterisk and
the database is completely
I have an Asterisk 13 installation with an E1 card and I thought
that DAHDI would be the default timing source for the system:
pbxcore*CLI> module show like timing
Module Description Use Count Status
Support Level
res_timing_dahdi.soDAHDI Timing
I had an old Asterisk installation die recently and we decided to
upgrade to Asterisk 13 to replace the old server. Everything seems to
be working with PJSIP but there is one issue. Asterisk talks to a
callmanager via a SIP trunk and send calls to PSTN (another country).
Most of the
I am getting flooded with these messages:
[Mar 1 12:25:29] WARNING[6962][C-005a]: chan_pjsip.c:712
chan_pjsip_write: Can't send 10 type frames with PJSIP
[Mar 1 12:25:30] WARNING[6962][C-005a]: chan_pjsip.c:712
chan_pjsip_write: Can't send 10 type frames with PJSIP
[Mar 1
On 2/24/16 12:10 PM, Aziz TestAccount wrote:
Hi All,
I'm looking for a PSTN Card that I can use with my Asterisk Server to
achieve the following goal :
1. Detect FAX signal and route it to a specific extension.
2. Detect an incoming call from the same PSTN line and route it to IVR.
Do
I am having a problem trying to compile
dahdi-linux-complete-2.11.0+2.11.0 on a CentOS 7.2 server. Version
2.10.2 compiles fine. Is there a new dependency for 2.11.0 that was not
required for previous versions? Here are some of the errors I get:
INSTALL
On 2/15/16 1:08 PM, Joshua Colp wrote:
Carlos Chavez wrote:
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport
section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along
On 2/15/16 12:50 PM, Joshua Colp wrote:
Carlos Chavez wrote:
Is it possible to use serveral protocols for a single transport section
in pjsip.con? In sip.conf you could use transport=udp,ws,wss so you
cound use webrtc along with your phones but if I try:
[transport-udp]
type=transport
protocol
Is it possible to use serveral protocols for a single transport
section in pjsip.con? In sip.conf you could use transport=udp,ws,wss so
you cound use webrtc along with your phones but if I try:
[transport-udp]
type=transport
protocol=udp,ws,wss
bind=0.0.0.0
I get an error that
On 2/11/16 12:36 PM, Joshua Colp wrote:
Carlos Chavez wrote:
I use realtime on my asterisk installation. I have always used mysql for
my realtime connection but as mysql seems to be on the "soon to be
deprecated" list of asterisk features I am trying to move to ODBC (still
using Mar
I use realtime on my asterisk installation. I have always used
mysql for my realtime connection but as mysql seems to be on the "soon
to be deprecated" list of asterisk features I am trying to move to ODBC
(still using MariaDB/Mysql on backend). I find ODBC support in Asterisk
very
I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I
keep getting this error:
[Feb 9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc:
Error in ExecDirect: -1, query is: INSERT INTO cdr
I am trying to port our Asterisk front end to Asterisk 13 but I
cannot get realtime static to work. Realtime for PJSIP, Voicemail and
Queues is working fine so I know res_odbc is configures properly. In
past versions of Asterisk I was using Mysql (res_config_mysql) to load
realtime
On 11/15/15 5:22 AM, Tzafrir Cohen wrote:
On Fri, Nov 13, 2015 at 04:01:33PM -0600, Carlos Chavez wrote:
I just purchased an Amfeltec USB-FXO adapter and am trying to compile
DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3. I have all the
dependencies but I get an error and cannot
I just purchased an Amfeltec USB-FXO adapter and am trying to
compile DAHDI 2.10 on a Raspberry PI running Pidora 2014 R3. I have all
the dependencies but I get an error and cannot finish. Is it even
possible to compile DAHDI for the ARM plataform? Here is the error I am
getting:
On 2015-09-24 17:08, Jeff LaCoursiere wrote:
Hi,
I have a client that has a 24 channel voice T1 that I have been using
e signalling over for a number of years. The local telco finally
got an ISDN switch and wants to move them to PRI. I didn't see this
as a big problem - I've done a few others
On 2015-09-13 10:16, Gokan Atmaca wrote:
Hello
I'm using the Fail2ban. I configuration below. I want to try to
prevent the continuous password. Fail2ban password that does not
prevent this form. (Asterisk 1.8 / Elastix interface)
What could be the problem ?
Asterisk log;
"Registration from
On 9/11/15 10:16 AM, Ethy H. Brito wrote:
Hi All
What, by definition, goes to the cdr table's "dst" column ??
In our setup, to get outside the user has to dial X before any number.
This goes to the dst with the X stripped out.
I recently made some changes in a macro and after that the X
On 9/11/15 12:59 PM, Jerry Geis wrote:
I have a setup where I have polycom phones in an office, behind firewall,
going out to a server located elsewhere. I have set
nat=force_rport,comedia for my phones.
so if I call OUT to my cell I get audio both ways and the call is fine.
My issue is if I
I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds fine.
The only problem I have is that when I call an internal SIP extension on
my PBX I do not hear the ring while I wait for the call to be answered.
My
On 9/9/15 4:22 PM, D'Arcy J.M. Cain wrote:
On Wed, 9 Sep 2015 16:11:03 -0500
Carlos Chavez <cur...@telecomabmex.com> wrote:
I am having a small problem that is driving me nuts. I can make
calls over my Webrtc client without any problems and audio sounds
fine. The only problem
On 8/26/15 1:15 PM, Tech Support wrote:
All;
I have a customer who is looking for a good speech to text
solution, either open source or reasonably priced commercial product,
I’m open to suggestions.
Thanks;
John V
For a commercial option try Lumenvox, had very good results. For free
On 2015-06-26 12:14, Tom Peters wrote:
Ok, commented out that line. It's still doing it. Reloaded dialplan.
Please don't tell me I have to restart asterisk.
asterisk -rx logger reload
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
dCAP #1349
+52 (55)9116-91161
--
On 2015-06-16 13:53, Steve Edwards wrote:
On Tue, 16 Jun 2015, sean darcy wrote:
There's no problem setting up vm on *. I can't use email off the
instance, since the assigned ip address doesn't have a PTR.
It looks too much like spam. The mail relays drop it.
Would configuring your own (or
On 2015-06-11 00:31, Luca Bertoncello wrote:
Hi list!
So, new day, new problem...
I tried right now to call from my cellphone a peer in my Asterisk.
The cellphone has correct credentials, but it's NOT registered on my
Asterisk, now.
I just tried to call a peer in my network, from a peer not
On 2015-06-06 03:19, s m wrote:
hello everyone,
i have question about fax detection on dahdi channels. does dahdi channels
detect fax and pass it? if yes, does it detects both types of fax (g711 pass
through and T.38)? finally, how can i enable it on dahdi_channels? i set
Ia had a server overload today because someone did a call forward
to their own extension. To do a call forward I write a key called CFWD
with the extensión number and number to dial . The main script tests if
the key/value exists and dials the number stored in the database. What
is an
On 5/29/15 1:16 PM, Ashwin Surendran wrote:
Hi,
I have multiple Asterisk servers in various parts of the world all
connected using dedicated VPN¹s.
Each of these servers have iax and dahdi TRUNK configured on them.
Occasionally the VPN¹s fail.
What I want to be able to do is on my dial plan,
We are having a strange problem today. We have a SIP trunk from a
provider and incoming calls are being dropped after the IVR when
attempting to connect to any internal phone. If a you dial a DID that
goes directly to a phone you can talk but the call will drop when you
attempt a
On 3/12/15 12:19 PM, Administrator TOOTAI wrote:
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: ${EXTEN} value is s
[...]
Daniel
On 3/11/15 12:48 PM, Salaheddine Elharit wrote:
hello list,
i use chanspy with the code below
[app-chanspy]
exten = _007.,1,Macro(user-callerid,)
exten = _007.,n,Answer
exten = _007.,n,Authenticate()
exten = _007.,n,ChanSpy(SIP/${EXTEN:3},dqs)
exten = _007.,n,Hangup
i have a question
On 2015-03-02 22:53, ricky gutierrez wrote:
Hi list , I have a question with account codes, all my outgoing calls
are authenticated, but now the boss wants to monitor these calls with
the codes.
example: maria has an extension 110, but peter was in place and use
the phone maria , maria then
On 3/2/15 3:23 PM, Marek Cervenka wrote:
hi,
is it possible use asterisk static realtime and config files
simultaneously in asterisk 11?
i want [globals] from extensions.conf in database, but dialplan in
extensions.conf config file
i saw this can be configured in stasis.conf in asterisk
I am having a problem with my queue_log. When an agent transfers a
call I am not getting the extension that was dialed for transfer, I am
only getting the name of the macro we use:
1425307308|1425307242.33367|PedidosKosmos|Agente
102|TRANSFER|s|macro-stdexten|13|52|1
Can anyone recommend a good WebRTC phone to use with Asterisk? I do
not mind if it is commercial or open source. Customers are starting to
ask for web solutions and we need to start testing.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161
--
I just finished installing Asterisk 13 on our test server and I can
now use PJSIP to register phones and make and receive calls. The only
problem I am having is that when I register multiple phones to a single
account only one of them rings. The AOR for the account has maxcontacts
at 3.
On 2014-10-02 12:32, Phil Ledon wrote:
We are trying to add voice mail to our hotel rooms. Our current phone
instruction cards say 'to reach voice mail dial ext 456. Replacing those
instructions is not feasible at the moment. We have Feature Code *97 that
takes them directly to their
On 9/23/14, 10:38 PM, Gokan Atmaca wrote:
Hello;
I was using the 1.8 version of Asterisk. However, due to a problem I
had to update. Update reporting system is broken when you have made.
Current version 11.10. I installed the modules in the system for
problems that are missing. I getting
On 9/22/14, 5:03 AM, Deepak Rawat wrote:
Hi,
We have a server with asterisk 1.4. We are upgrading to asterisk 12.4.
Is there a way to install 12.4 on the same machine? At any point we
will only run either 1.4 or 12.4.
The answer is that you can but you really shloud not as it
On 2014-09-15 16:45, Gao wrote:
Hi,
I am using this dialplan to record incoming calls:
.
exten = 3331122,n,Set(MONITOR_FILE=${RECDIR}/${UNIQUEID})
exten = 3331122,n,MixMonitor(${MONITOR_FILE}.wav,b)
exten = 3331122,n,GoSub(stdexten(${Ext1007}))
exten = 3331122,n,Voicemail(1007@default,)
On 9/5/2014 2:18 AM, Horace Miles wrote:
Hello everyone, my name is Miles, I am fairly new to asterisk. I have
recently begun to learn asterisk and I have a couple of questions.
1. After installing asterisk using the following instructions;
a.sudo mkdir /usr/src/asterisk amp;amp; cd
We are currently migrating from a Nortel pbx to Asterisk and we
have been able to convert most of the functions that people are used to
but there is one I have no clear idea how to do. The scenario is:
Boss calls secretary from outside the office to get connected to
another outside
On 8/25/14, 11:44 AM, Joshua Colp wrote:
On 8/25/2014 2:36 AM, Brian LaVallee wrote:
Hello,
Here's a fun issue that recently caused me some serious heartache.
Hope this helps others from making the same mistake.
Did you know that the configuration parser supports block-comments.
Like an
On 8/13/14, 11:31 AM, Matthew Jordan wrote:
On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Is anyone using asterisk on CentOS 7?
If so, is it working fine and as expected?
Random data point: the Asterisk project's build agents are still on CentOS 6.
Your
I am having a very strange problem. We use Asterisk 11.X (have
tried several versions, including certified) which reads its config
files in realtime from a SQLITE3 database. Everything runs fine but
lately asterisk has been crashing when we issue a reload command via
Manager. Our web
On 6/3/14, 11:43 AM, Stefan Gofferje wrote:
On 06/03/2014 06:06 PM, Eric Wieling wrote:
Have you tried RetryDial()?
I want it to be a conscious decision and not just automatically in every
call. For the vast majority of my call I can just try some time later
but some people I need to get a
I just finished migrating our web interface from Mysql to SQlite3
and everything seems to be working fine. I just have one detail. The
following keeps appearing on my logs:
[Apr 29 13:09:32] WARNING[30494]: res_config_sqlite3.c:520
realtime_sqlite3_execute_handle: Could not execute
I have found Asterisk using only SIP is very responsive on virtual
machines. We have used VMs for call center applications and for complex
IVR solutions without problems. Obviously there is overhead running a
VM so you can never expect a VM to perform as well as bare metal.
Running a
I am using Realtime on Asterisk 11.5 with a SQLite3 backend. While
everything seems to be working fine I keep getting this error on my log
files:
[2014-02-17 19:55:18] WARNING[20569] res_config_sqlite3.c: Could not
execute 'UPDATE sip_buddies SET ipaddr = '192.168.2.23', port =
'5060',
I have a customer with a more or less unique need. Right now we
are using Wombat as a dialer software so they can contact clients for QA
purposes. Everything is working very well and their contact center
productivity is way up from the old manual dialing method.
The only thing we
On 10/31/13, 8:44 AM, Rizwan Hisham wrote:
Hi all,
Is there any way of originating calls in future without using call files?
We have 2 servers (1 active at a time). If we use call files with
modification date in future, on the 1st server and it dies and, we
activate the second server but we
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On 9/24/13 4:04 PM, Tim Nelson wrote:
Greetings-
I have an odd scenario where I need to dial an extension (lets call
it 555), the system prompts for a list of voicemail boxes, then
once complete, allows the caller to leave a voicemail that is
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On 8/1/13 9:17 PM, Michael L. Young wrote:
- Original Message -
From: Carlos Chavez cur...@telecomabmex.com To:
asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013
8:41:19 PM Subject: [asterisk-users] External sip phones
and it is the same as in past versions. externaddr and localnet
are set to the proper values. Any ideas?
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
--
_
-- Bandwidth
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I have an Asterisk 11.4 SIP only system. We are using a SIP trunk
for outside calls. We are having a problem with calls dropping after
a transfer.
Outside call awswered by phone 101
101 transfers to 100 (attended transfer)
call is dropped
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I have been struggling with an audio issue for a week now and have
not been able to solve it.
We have an Asterisk server (now running 11.4 but started with 1.8)
with several sip phones on an internal network and a SIP trunk for
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The best guide is the Asterisk Fefinitive guide and a virtual machine
so you can install several Asterisk servers and make them talk to each
other.
On 6/7/13 1:20 PM, Michael Gilleran wrote:
Greetings. Anyone have any recommendations for studying
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The free route is DIY. A simple script in your favourite scripting
language will do nicely. If you need things to control the dialer and
have statistics I would go with something like Wombat Dialer from the
makers of Queuemetrics. It is free for
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On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez
cur...@telecomabmex.com mailto:cur
cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez
cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote:
On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu
@oficina:1] Macro(SIP/2003-000e,
stdexten,2001,SIP/2001) in new stack
-- Executing [s@macro-stdexten:1] NoOp(SIP/2003-000e,
LLamada a extension estandar 2001) in new stack
-- Executing [s@macro-stdexten:2] NoOp(SIP/2003-000e,
LLamada desde: Carlos Chavez 2003) in new stack
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I am trying to set the CDR(userfield) to a certain vaule using the
application map of features.conf but I am not able to do it. When I
receive a call I would like to tag it with a client code (3 digit
numeric) so I can referenci it later from
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On 1/7/13 6:53 PM, Jerry Geis wrote:
According to this:
https://wiki.asterisk.org/wiki/display/AST/Video+Telephony yes.
I have a local server with two video phones - running SIP to each
phone. Works. Then I have an IAX2 connection from
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On 11/24/12 4:07 PM, Richard Kenner wrote:
I have a peculiar RTP issue. I'm experimenting with Jitsi as a
softphone on one of my desktop Windows machines. That machine can
either be connected to Asterisk via an VPN connection (with a
static IP
I have a new install and the customer is complaining that they hear
noise on all calls, no matter if it is internal or external, desk phones
or softphones. The noise is only present when the user is speaking, not
the remote side. The remote side does not hear the noise, only the
local
On 11/13/12 4:31 PM, Mark Engelhardt wrote:
Carlos,
I think the noise you are hearing might echo cancelation that is broken or set
incorrectly. Maybe the card and asterisk are both trying to echo cancel?
Mark
On Nov 13, 2012, at 1:52 PM, Carlos Chavez wrote:
I have a new install
On Fri, 2012-10-05 at 05:21 -0700, Vieri wrote:
--- On Fri, 10/5/12, Vieri rentor...@yahoo.com wrote:
An Asterisk queue uses field names / config variables such
as:
announce-holdtime
However, documentation regarding realtime is very unclear.
voip-info.org suggests to use
On Fri, 2012-08-31 at 17:53 +, Giuseppe Longo wrote:
Hi,
has anyone tried asterisk on arm processors? how is the performance?
have encountered problems in the compilation?
Thanks,
Regards.
I have installed Asterisk on a Raspberry Pi and it works very well for
a small site.
On Tue, 2012-08-14 at 18:43 +0100, Goke M Aruna wrote:
hello all,
I have call queue management system where all call comes in, put in
the queue while the caller speak with the online support team /
teacher.
However, my major concern is those under MOH (in the queue) will not
be able to
Look for AMD (Answering machine detection).
On Fri, 2012-08-03 at 14:42 -0700, sathiish kumar wrote:
I am looking for ways to detect if there is some person talking on the
other side of the line and trigger some events based on that.. is
there any possible way through which this could
I have a SIP line that is working fine when I make calls from IP
phones. I can send and receive calls. The problem is that if I try to
dial from the CLI using the originate command or use an AMI connection
to originate a call I get the following error:
originate
On Wed, 2012-07-25 at 16:05 -0400, Ken D'Ambrosio wrote:
Hi, all. I see that, with Asterisk 10, there've been some additions with an
eye toward conferencing, and, apparently, hooks for video conferencing.
Googling like crazy, however, has given me little to go on. I've been tasked
with
This is as easy as running an AGI on your 911 rule to do whatever you
want. The AGI can dial multiple phones, send emails, page you, etc.
Even without the AGI you can do many things from the dialplan.
On Sat, 2012-06-09 at 07:51 -0600, Nunya Biznatch wrote:
Can you set up asterisk so
Yesterday a customer was attacked from the following IP addresses so
add them to your blacklist:
iptables -A INPUT -s 37.8.119.75 -j DROP
iptables -A INPUT -s 37.8.22.240 -j DROP
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
/listinfo/asterisk-users
--
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001
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