gt; #5,peer/caller,Macro(RaiseHand)
extensions.ael
Set(DYNAMIC_FEATURES=RaiseHand);
MeetMe(1234,F);
I have tried with and without the F parameter...
Any suggestion?
Leandro
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I just noticed there is some sort of new spandsp library.
http://www.soft-switch.org/downloads/spandsp/snapshots/
The version reported was still 0.0.6 and there is absolutely no "whats new"
file.
Is there anyone with more details
Hello,
am I wrong or the audio file for vm-rec-name in en_GB package says "pound"
instead of "hash"?
Pound should be for American while British use hash for the # key.
Leandro
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_
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Unfortunately the only log messages regarding that channel are the "joined"
and the "left" for both legs.
VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
VERBOSE[18779][C-066c] bridge_cha
provider side.
Leandro
2016-09-15 19:18 GMT+02:00 Max Grobecker :
> Maybe the client just put the call on hold.
> So the call technically has not ended AND the client does not need to send
> or handle any RTP data.
> Is there any mention of "music on hold" for this channel
ge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:28] VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
Any idea?
Leandro
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users. I'd like to have
this setting different for each Music on Hold class.
Is it possible?
Leandro
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Join the Asterisk Community at the 13th Ast
No. I thank you for all the hard work done and dedication to the project.
Leandro
Il 06/Lug/2016 11:10 PM, "Joshua Colp" ha scritto:
> Leandro Dardini wrote:
>
>> This is a great news, thank you. I have open the issue,
>> https://issues.asterisk.org/jira/browse/
This is a great news, thank you. I have open the issue,
https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the
relevant files, let me know if you need more info.
Leandro
2016-07-06 21:46 GMT+02:00 Joshua Colp :
> Leandro Dardini wrote:
>
>> Hello,
>> I'd li
p, it should be completely
removed.
Leandro
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the
pjsip extension has registered to?
Leandro
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Hi,
I'd like to record the barged call... but whichever leg of the call I try
to barge, my speaking is never recorded using MixMonitor. Any idea about
the reason?
Leandro
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I run in a weird issue with a BLF application I have written... this
application is just receiving events from Asterisk Manager Interface and
blink the lights accordingly. All almost work perfectly, except when a
pickupexen is used when multiple extensions are dialed.
If extension 105 dials extens
Which operating system are you using? I have experienced the same problem
on several OS except for CentOS 6. I suppose an ODBC problem on newer OS
version.
Leandro
Il 24/Feb/2016 05:30 PM, "Maxime" ha scritto:
> Dear list,
>
> i have a issue
>
> Asterisk crash (Modul
Please chech also MiRTA PBX http://www.mirtapbx.com ... it is a multitenant
realtime multiserver interface.
Leandro
Il 23/Dic/2015 09:06 AM, "er ic" ha scritto:
> Although, I do like the OS information. I personally am a fan of CentOS.
>
> I realize now that the platform wa
I see, really thank you ... I have just migrated my config. By the way ...
is pjsip realtime supporting realtime registrations?
Leandro
2015-09-08 21:23 GMT+02:00 Joshua Colp :
> On 15-09-08 04:21 PM, Leandro Dardini wrote:
>
>> I have some problem finding a smart way to add inbou
with them?
Leandro
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asterisk-users ma
@devel.mirtapbx.com;transport=UDP>,sip:1...@devel.mirtapbx.com;transport=UDP,102,103,Second
test 4,)=SUCCESS") in new stack
Asterisk interprets the first "=" as assignment. In the debug log I found:
Variable: ODBC_LOG_SMS(1,ONNET,,sip:1...@devel.mirtapbx.com;transport=UDP,1
{MIXMONITOR_FILENAME})exten => s,n,StopMixMonitor()
[macro-unpause-recording]exten => s,1,NoOp(Resuming Recording -
MIXMONITOR_FILENAME is ${MIXMONITOR_FILENAME})exten =>
s,n,MixMonitor(${MIXMONITOR_FILENAME},ab)ldardiniNewsterisk Leandro
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___
correctly, but then, when the SIP SUBSCRIBE arrives, the mailbox is not
found. If I run a SIP SHOW PEER, the peer is shown without the mailbox.
Have you ever noticed a similar behavior?
Leandro
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peer (just to
understand, if I run "sip show peer 104-TEST", I see the Mailbox empty. If
I run the "sip show subscriptiona", I don't see any subscription for the
MWI but only for the BLF.
Is there anyone facing the same problem? Ho
The HASH function is really useful when you have to deal with values loaded
using func_odbc, but how do you use with the LOCAL function? Is it possible
to define a HASH as LOCAL?
Leandro
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.
I just need to pass a variable from the channel placing the call to the
followme to the channel where the extension is dialed by followme. Any idea?
Leandro
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when the first
answers, the other stops ringing.
Any idea to make the first continue to ring until the other accept the call?
Leandro
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call comes from outside.
The bad CallerID is displayed only on Cisco 504G phones and it is
transmitted as a Remote-Party-ID
Is there anyone else also getting this bad behavior?
Leandro
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has moved
in another context, then the new call will be started from such context
with unpredictable results.
Do you have any idea to make all transfers to be applied to the context
defined in the sip.conf instead of the context where the call is running in
that moment
Hello,
I am struggling with what seems a common unresolved problem, changing the
password from voicemailman when using a realtime engine (adaptive_odbc in
my case, connected to mysql).
I have seen messages dating back to 2007 with this problem and the last one
was bug 5168, reported as closed, but
The output of the "Sip show subscriptions" is a formatted text with columns
cut to fit in the "page". It can be better than nothing, but I really
dislike to parse it and show incomplete data.
Leandro
2015-01-16 0:03 GMT+01:00 Alex Epshteyn :
> You can use "Com
Hello,
almost any useful CLI command has an analogue on Asterisk Manager
Interface, but I cannot find a way to get the list of subscriptions using
AMI. Which is the command, if any? The CLI command is "sip show
subscriptions"
00 - Call completed
elsewhere is sent over the channel for 104, but that is not transmitted to
106.
Is it a way to make it happen?
Leandro
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Hello,
do you know if it is possible to define the SLA configuration in the
database for realtime usage with asterisk?
Leandro
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(Alert-Info: Ring Answer);
SIPAddHeader(P-Auto-Answer: normal);
Any other ideas?
Leandro
PS
I have set the "Auto Answer Page" to yes
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New to Asteris
more for having MWI to work on asterisk 12.6? I
just moved the configuration used for asterisk 12.3 to the one running
asterisk 12.6
Leandro
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New to Ast
Hello,
have you noticed the message num (VM_MSGNUM) is off by one?
For example, I receive the following message:
"Just wanted to let you know you were just left a 0:03 long message (number
7)"
but in attach there is the msg0006.wa
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik :
> Do the pause/unpause in a Macro or Gosub and reference that from the
> features.conf
>
> Also, make sure you put the filename into a variable and give it full
> inheritance so you can resume recording t
Hello,
I have a recording started in the dialplan with the MixMonitor application.
I want to be able to stop it during a call and maybe restart it.
I tried using the value defined in [featuremap] but it starts another
MixMonitor application even if there already one instead of stopping it.
Any id
Hello,
I have my provider dropping the calls after 41 seconds of not receiving any
RTP from my asterisk. Obviously there is no RTP back when the caller is
leaving a message in the voicemail. Is it possible to have asterisk
generate some RTP packet back?
Leandro
It is the way it works. First the phone sends a REGISTER without any
password. Asterisk answers with a "Unauthorized" and provide a nonce to be
used for the next registration attempt, using it to encrypt the password.
Leandro
2014-05-14 13:12 GMT+02:00 Olli Heiskanen :
>
> He
e Local
channel used by asterisk to place the new call, the originating extension.
In the logs asterisk says "Thanks to SIP/104-DEVEL..." but in which
variable can I find this value?
Leandro
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... maybe it is just someone trying to place some free calls
Leandro
2014-02-12 19:05 GMT+01:00 Mike Diehl :
> Hi all,
>
> I've got a customer who's reporting "ghost calls." Essentially, the phone
> rings, they pick up, and there's no body there.
>
> It is N
very
long timeout of 3600 seconds.
Leandro
2014-02-06 17:18 GMT+01:00 Mike Diehl :
> Hi all,
>
> I have an SPA112 that in sitting behind a Ubee cable modem. The internet
> link is solid, but the device becomes unreachable within a day or so of
> being rebooted. Then the customer
I love you all
:-)
Leandro
2014-02-05 Richard Mudgett :
>
>
>
> On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini wrote:
>
>> Hello,
>> I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
>> the ${CDR(start)} is not returning any data. Other
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems the
${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
correct values. Where is my mistake? Has this function being renamed?
Leandro
I have converted the normal Park application and I can only alert you about
the syntax change. I suspect also in the ParkAndAnnounce command, the
parameters are ordered completely different.
Leandro
2014-01-30 Anders Larsson :
> Hi
>
> I'm trying to get the rebuilt parking fu
2014/1/23 Matthew Jordan
> On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini
> wrote:
> > When you use a product which version number is 11 or even 12, you might
> go
> > with the assumption all big bugs are fixed and then you find there is a
> > huge, important, exp
from extension 100
to extension 101 lasting 10 seconds. What about the 100 seconds call from
100 to 00VERYEXPENSIVEDESTINATION? It will never get billed.
How do you manage these cases?
Leandro
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Please paste the actual code. First has to be the Wait and then any other
thing.
Leandro
2014/1/21 Jakob-Matthias Böttger
> i already added a Progess() and Wait(5) and it still does not detect
> faxes.
>
>
> Am 21.01.2014 16:53, schrieb Leandro Dardini:
>
> I am not s
I am not sure, but try to add a wait(2) as first command. When I want fax
detection, I insert always a small delay for letting the fax detection
routine to detect it.
Leandro
2014/1/21 Jakob-Matthias Böttger
> Hi
>
> The log i've posted
>
>
> == Using SIP VIDEO CoS
I am going to try a Lync server/asterisk integration, so I really
appreciate!
Leandro
2014/1/21 Lincoln King-Cliby
> Ok, so now I just feel kind of stupid. After I got home I decided to play
> with this a little more.
>
>
>
> After far too long I realized that part of the
It is really more interesting the receiving part. Can you paste here?
Leandro
2014/1/21 Jakob-Matthias Böttger
> Hello everybody
>
> I'm trying to enable the Digium res_fax app at my *11.7 Server.
>
> a fax show stats comes up with
> FAX Statistics:
> --
Yes, thank you. Maybe I have found the problem. The asterisk server is
behind a nat and the RTP port range was not redirected to the asterisk box,
so the Symmetric RTP cannot work because the asterisk is not receiving any
RTP packet from the remote phone.
Leandro
2014/1/16 Ishfaq Malik
>
o yes, but asterisk seems ignoring them.
Force rport : Yes
Symmetric RTP: Yes
Asterisk is behind a nat the the externip and localnet has been configured.
The local net on the asterisk network is different from the local net on
phone.
What else could I check
Just use VNC...
2013/12/20 Goke M Aruna
> Thanks AJ,
> The capturing of agent activities on their desktop by the supervisor.
> Regards
> On 20 Dec 2013 12:18, "A J Stiles" wrote:
>
>> On Friday 20 December 2013, Goke M Aruna wrote:
>> > Thank you AJ,
>> > Just want to know from people who uses
nd I'll be happy to help you
Leandro
2013/12/11 Mario Giammarco
> Hello,
> I need to setup this configuration:
>
> - asterisk as IVR;
> - dect phones.
>
> So basically I need a "standard set" of features:
>
> - each dect phone has its extension so I
Hello friends,
when a call arrives in the queue, a CDR record is created, but there is no
info about which agent has picked up the call. I can find that info only in
queue_log.
Is there a way to have that info in the CDR or maybe in a variable in the
"h" context, when the call is ended
On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?
Leandro
2013/11/25 Daniel - Asterisk
> Hello Friends:
>
> I've just installed Asterisk 11 on my Linux (debian) server but it is not
> starting up when trying with "asterisk -vvv
20:09:01] NOTICE[12287][C-0001689e]: chan_sip.c:22914
handle_response_invite: Failed to authenticate on INVITE to '"Leandro
Dardini" ;tag=as1c0d8470'
-- SIP/78.11.22.33-000144c3 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Which is the correct synta
quot;orders" in the list of results, so
the members for the queue are returned in random order.
Anyone experiencing the same problem? How do you solve it?
Leandro
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It seems very good! I am going to test it when I have a bit of time!
Leandro
2013/11/14 Ryan Wagoner
> I haven't tried it, but the res_corosync module states it will sync device
> state across servers.
>
> https://wiki.asterisk.org/wiki/display/AST/Corosync
>
>
> On
out the change, so both
asterisk are taken aligned.
Let me know if you need additional details.
Leandro
2013/11/13 Lincoln King-Cliby
> Hi All,
>
>
>
> We’ve been running Asterisk for years in our offices but just recently
> replaced an Asterisk Appliance* in our smaller off
2604
>
> Cell: 973.390.1090
>
> www.xaccel.net
>
>
>
>
>
>
> *CONFIDENTIALITY NOTICE: This e-mail message, including any attachments,
> is for the sole use of the intended recipient(s) and may contain
> confidential and privileged information which should
the transmission of this information back to
the caller. How can I do it?
I tried setting
Set(CONNECTEDLINE(num-pres)=prohib);
but it doesn't seem to sort any effect.
Where am I wrong?
Leandro
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In my dialplan I'd like to send a "603 Declined" message to the user
placing the call. I see the commands for the Busy and Congestion, but not
the one for the Declined. Any help?
Leandro
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Again, the authenticate function can help you
Leandro
2013/5/20 Felix Vazquez
> How do I make a user dial a passcode if he wants to make an
> international call?
>
>
>
> --
>
> This electronic message contains information from BOSH Global
I think it can be worth checking the authenticate function.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate
2013/5/20 Felix Vazquez
> How do I make a user dial a passcode to make calls through asterisk?
>
> We would like to place a phone at a client’s location for our employee b
Is the "echo" application suitable to you?
Leandro
2013/5/20 CDR
> Dear friends
> I need to loopback the audio on my channel. Did anybody on the development
> team thought about a function or app that would do that? If it is not
> clear, I mean that whatever audio I get,
Uhm ... I see the easy way will be to tcpdump the connection between the
asterisk and the mysql database server and to dump the exact SQL syntax
used. It will be something wrong...
Leandro
PS
tcpdump -i any -n -s 1500 -w /tmp/data.pcap port 3306
2013/4/18 Tommy Cooper
> Thank you for y
You need a "name" column. This is my queue table:
CREATE TABLE IF NOT EXISTS `queue` (
`name` varchar(128) NOT NULL,
`musiconhold` varchar(128) DEFAULT NULL,
`announce` varchar(128) DEFAULT NULL,
`context` varchar(128) DEFAULT NULL,
`timeout` int(11) DEFAULT NULL,
`monitor_join` tinyin
You are right for the commands to prune and clear the cache. But what is
the meaning of the meaning of the configuration parameter rtautoclear if it
is not clearing the cache?
Leandro
I am typing from my mobile phone...
Il giorno 26/mar/2013 14:38, "Michael L. Young"
The phone will renew the registration before it expires, so maybe it
never "expires".
I have tried to set the rtautoclear to 60, but the result is the same,
the new password is never enforced.
Any suggestion apart from removing the rtcachefrie
I dont apply any secret recipe while installing asterisk, but maybe you can
share yours...
I am typing from my mobile phone...
Il giorno 23/mar/2013 14:34, "Nick Khamis" ha scritto:
> Hello Everyone,
>
> We are getting some rather poor results (relative) with our Asterisk
> setup. Not sure if we
is important to know the reason the call is disconnected.
Start checking who is sending the BYE and if before the BYE there is
other weird packets, like retry of packet sending ...
A simple "tcpdump" can help explain all the mistery.
Leandro
--
__
You can add custom fields in the CDR, so your dialplan can store start
time, end time and duration whenever you like.
Just use something like the
Set(CDR(customfield)=100);
Leandro
2013/3/18 RSCL Mumbai :
> Thank you every one.
> Now I understand why I was confused.
> I have always b
ered and the end of the call.
In your example, duration and billsec will differ for just a second,
the time from the "Call Connected to asterisk" and the "Welcome
greeting starts".
Leandro
2013/3/18 RSCL Mumbai :
> I am using SIP.
>
> I am still a bit confused about
s it a problem of
codec? Is it a problem of license?
Leandro
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s a
max load of 0.5 with 24 cores. When I was using a 4 cores server with the
same number of channels, I get a load of 3 ... so the load x core relation
is valid. I think it will be good to have a load not over 4 for a 4 core
server, so you can have at least 200 active channels on the server. If you
a
If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.
Leandro
I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, "Luis H. Forchesatto" <
luisforchesa...@gmail.com> ha scritto:
> Greetings.
>
> I got an
all the load on a single server, but to spread the phone among multiple
servers. The best will be to have multiple asterisks working together using
realtime extensions. It is not difficult to make.
Leandro
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g), average usage of conference call and other audio mix feature,
reaching a max of 60 CPS and an average of 150 channels without problems.
The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works
fine even on the old
I think a simple tcpdump of the traffic will show the mystery. It can be
your provider doing something nasty. Have you tried using some other cheap
SIP termination? or arrange a fake termination yourself on another server?
Leandro
2013/3/1 Gerard
> I thought it was the re-invites too, bu
call to
Bob at ext 300, then Bob will see the callerid 200 on his phone. That is
not true if the dial is made inside a Macro. In this way, Bob will see
s
The macro can be something as simple as:
macro dialpeer(number) {
dial(SIP/number);
}
Leandro
2013/2/24 Mitul Limbani
>
lay.
I am just rewriting all the dialplan getting rid of the macro and using
gosub, even if asterisk is complaining about "application call to gosub
affects flow of control, and needs to be re-written using AEL if, while,
goto, etc. keywords instead!", but I am not seeing a
The h exten is triggered when the channel is hangup, so you cannot send any
voice data on it.
Leandro
2013/2/21 Enrico Pasqualotto
> Yes, correct now it works for Dial.
> I think is the same with "c" option on Queue, do you think there's a way
> to do it on h exten?
&
>
> Thanks
>
> Enrico.
>
>
If you choose to go with the Dial command and use the "g" option, you have
not to use the "h" extension, but just provide a next priority. Your
dialplan has to be:
[from-test]
exten => _X.,1,Dial(SIP/${EXTEN},60,rjtTg)
*exten =
>
> Regards
> Akhilesh
>
>
I am sorry if I haven't completely understood your question, but english is
not my native language. If calls from server_A and server_B are put in the
same queue in server_X, how can one of them b
e media directly to phone B to save
bandwidth. It is named "reinvite"
Leandro
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Check if you have selinux enforcing anf try to disable it
I am typing from my mobile phone...
Il giorno 04/feb/2013 18:43, "C. Savinovich"
ha scritto:
>
> I would just type in the web service url manually in a browser, and if the
> browser displays the response, then there it is, the connection
--
> > Carlos Alvarez
> > TelEvolve
> > 602-889-3003
> >
> >
> This is so true!
>
>
If you have no NAT or dynamic IP in your network, you can just remove the
registration process and assign to each peer its IP address.
Leandro
--
__
You can just shorten the time the phone device register on the asterisk
server. It is up to the peer to send the registration command. It cannot be
triggered or forced in any way.
Leandro
2013/1/30 XBrian
> I am aware that the direction is from peer to asterisk. Its
> a valid question
heck the content of the SIP packet containing the
message. That way you'll know if the asterisk or the softphone is to blame.
Leandro
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New to Asterisk?
The simplest way is to use the Random function and to pickup one number
from 1 to 3 and use that line.
Leandro
I am typing from my mobile phone...
Il giorno 29/gen/2013 11:35, "Salaheddine Elharit" <
salah.elharit...@gmail.com> ha scritto:
> I am installing asterisk 1.4 wi
how to do it
using elastix interface. Maybe you can have more luck asking to some
elastix related mailing list.
Leandro
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It is a shame we were unable to find the solution to your problem. Do you
want to setup a test system like the good one and let me access it to check
what is going on? I am really really curious.
Leandro
Il giorno 26/gen/2013 19:49, "Dan Journo" ha
scritto:
> > It is really un
-
> _____
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-user
a tcpdump on secondary server to see if for some obscure reason the
phones try to contact the secondary asterisk?
Leandro
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New to Asterisk? Joi
15.0. I'll upgrade it to 1.8.20.1 when I can and see if it makes a
> difference.
>
> --
>
I am curious, is your version of asterisk correctly compiling the regserver
field? Each server needs to have a distinct server name.
Leandro
--
_
>
> Its probably an issue with the version of Asterisk we are using because I
> haven't had this problem in the past.
>
>
I am running the latest 1.8 version. Which version are you running?
Leandro
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--
2013/1/23 Dan Journo
> > We have never experienced that and use realtime with multiple asterisk
> servers.
>
> We've only recently started seeing the problem.
>
> To simplify the issue, assuming we have two servers, Asterisk1 and
> Asterisk2...
>
> Asterisk1 is a primary server and Asterisk2 is a
>
> T: 0161 820 8353
>
>
>
All depends by the number of sip peers and the number of addition/deletion
you make. If you have static files, you have to "sip reload" every time you
add/rem
Can you please post a dialplan excerpt about using these variables. I just
tried using them, but they are all empty. Maybe I am making the same
mistake of you.
Leandro
2013/1/22 Administrator TOOTAI
> Please forget this message, BLINDTRANSFER is working, I had a typo in the
> dialpla
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