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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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]
on
the provider side.
Leandro
2016-09-15 19:18 GMT+02:00 Max Grobecker <max.grobec...@ml.grobecker.info>:
> 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
96f37260>
[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 AstriCon
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" <jc...@digium.com> ha scritto:
> Leandro Dardini wrote:
>
>> This is a great news, thank you. I have open the issue,
>> https://issues.aster
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 <jc...@digium.com>:
> Leandro Dardini wrote:
>
>>
it should be completely
removed.
Leandro
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aste
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
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" <mcaillet...@openip.fr> ha scritto:
> Dear list,
>
> i have a issue
&
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" <email.eherr9...@gmail.com> ha scritto:
> Although, I do like the OS information. I personally am a fan of CentOS
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 <jc...@digium.com>:
> On 15-09-08 04:21 PM, Leandro Dardini wrote:
>
>> I have some problem finding
with them?
Leandro
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asterisk-users mailing
BC function is not executed.
Is there a way, beside using REPLACE, to avoid this problem?
Leandro
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is ${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
, 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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(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? How have you solved it?
leandro
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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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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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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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?
Leandro
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,
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 a...@thirdlane.com:
You can use Command command, and sip
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
Leandro
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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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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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.wav
Leandro
Can you post an example?
Leandro
2014-08-28 0:47 GMT+02:00 Ishfaq Malik i...@pack-net.co.uk:
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
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
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 ohjelmistoarkkite
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 mdiehlena...@gmail.com:
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 NOT one-way audio, and it doesn't
timeout of 3600 seconds.
Leandro
2014-02-06 17:18 GMT+01:00 Mike Diehl mdiehlena...@gmail.com:
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 goes
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 love you all
:-)
Leandro
2014-02-05 Richard Mudgett rmudg...@digium.com:
On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.comwrote:
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
the ${CDR(start)} is not returning any data. Other
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 aster...@adev.se:
Hi
I'm trying to get the rebuilt parking
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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2014/1/23 Matthew Jordan mjor...@digium.com
On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini ldard...@gmail.com
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
It is really more interesting the receiving part. Can you paste here?
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
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:
---
Current Sessions
I am going to try a Lync server/asterisk integration, so I really
appreciate!
Leandro
2014/1/21 Lincoln King-Cliby linc...@controlworks.com
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
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 ja...@j-mb.de
Hi
The log i've posted
== Using SIP VIDEO CoS mark 6
== Using
Please paste the actual code. First has to be the Wait and then any other
thing.
Leandro
2014/1/21 Jakob-Matthias Böttger ja...@j-mb.de
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 sure, but try
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 i
, 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?
Leandro
Just use VNC...
2013/12/20 Goke M Aruna gok...@gmail.com
Thanks AJ,
The capturing of agent activities on their desktop by the supervisor.
Regards
On 20 Dec 2013 12:18, A J Stiles asterisk_l...@earthshod.co.uk wrote:
On Friday 20 December 2013, Goke M Aruna wrote:
Thank you AJ,
Just
be happy to help you
Leandro
2013/12/11 Mario Giammarco mgiamma...@gmail.com
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 can call it directly;
- handover of a call
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?
Leandro
On which kind of processor are you trying to run asterisk? Is it a real or
emulated CPU?
Leandro
2013/11/25 Daniel - Asterisk earohua...@gmail.com
Hello Friends:
I've just installed Asterisk 11 on my Linux (debian) server but it is not
starting up when trying with asterisk -vvc
-0001689e]: chan_sip.c:22914
handle_response_invite: Failed to authenticate on INVITE to 'Leandro
Dardini sip:100@91.11.22.33;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 syntax to use to dial directly
asterisk are taken aligned.
Let me know if you need additional details.
Leandro
2013/11/13 Lincoln King-Cliby linc...@controlworks.com
Hi All,
We’ve been running Asterisk for years in our offices but just recently
replaced an Asterisk Appliance* in our smaller office with an actual
server
It seems very good! I am going to test it when I have a bit of time!
Leandro
2013/11/14 Ryan Wagoner rswago...@gmail.com
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 Thu, Nov 14
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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Do you have compiled asterisk by yourself? In the Voicemail Build Option,
what option have you selected? I think you need to select ODBC Storage
and then configure ODBC on the system to connect to your database.
Leandro
image003.png
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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Is the echo application suitable to you?
Leandro
2013/5/20 CDR vene...@gmail.com
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, I send
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 felix.vazq...@theboshgroup.com
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
Again, the authenticate function can help you
Leandro
2013/5/20 Felix Vazquez felix.vazq...@theboshgroup.com
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
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`
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 tomcoope...@yahoo.com
.
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 rtcachefriends?
Leandro
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 myo...@acsacc.com ha
scritto
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 sym...@gmail.com ha scritto:
Hello Everyone,
We are getting some rather poor results (relative) with our Asterisk
setup. Not
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
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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 rscl.mum...@gmail.com:
I am using SIP.
I am still a bit confused about answered billed time.
For example:
00:00 -- Call Connected to asterisk
00:01
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 rscl.mum...@gmail.com:
Thank you every one.
Now I understand why I was confused.
I have
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
accept more load, then you can get more channels.
Leandro
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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 hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz
Leandro
. It is not difficult to make.
Leandro
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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 extension on my Elastix who cannot pick
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 gsara...@rarcoa.com
I thought it was the re
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 any other way...
Leandro
the 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 mi
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 being abandoned? Calls will be
processed in the same order as they arrive.
Leandro
)*
[play]
exten = s,1,Noop(play)
exten = s,2,Saydigits(123579)
Leandro
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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 e.pasqualo...@netspin.it
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?
My goal
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 c.savinov...@itntelecom.com
ha scritto:
I would just type in the web service url manually in a browser, and if the
browser displays the response, then there
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
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New to Asterisk
?
Is there a variable to be set?
Any ideas will be most welcome
If I was in your shoes (is this the right English sentence?) I'll run a
tcpdump command to check the content of the SIP packet containing the
message. That way you'll know if the asterisk or the softphone is to blame.
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 bobo...@yahoo.co.uk
I am aware that the direction is from peer to asterisk. Its
a valid
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 with 2 ISP and i have one
delivered calls inspite
of Level-1 agents being available.
Any help or pointers are appreciated.
Thx,
Vai
I know for sure how to do it in asterisk, but I don't know how to do it
using elastix interface. Maybe you can have more luck asking to some
elastix related mailing list.
Leandro
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 d...@keshercommunications.com ha
scritto
and it is a berkely db, but other database backends
seem available. Are you sharing also this database between the two servers?
It is the only option left...
Leandro
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to see if for some obscure reason the
phones try to contact the secondary asterisk?
Leandro
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you make. If you have static files, you have to sip reload every time you
add/remove a peer. With realtime is all realtime. I have switched to
realtime peers some times ago with great benefit.
Leandro
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2013/1/23 Dan Journo d...@keshercommunications.com
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
running the latest 1.8 version. Which version are you running?
Leandro
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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
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-- Bandwidth
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 ad...@tootai.net
Please forget this message, BLINDTRANSFER is working, I had a typo
call).
Leandro
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