Hello,
I am trying to use a dynamic_features during a MeetMe conference without
any luck. The dynamic_features defined macro works great during a normal
call, but is ignored while on a MeetMe conference.
extensions.conf
[macro-RaiseHand]
exten => s,1,DumpChan(1)
features.conf
RaiseHand => #5,peer
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?
Leandro
--
_
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
--
_
-- Bandwidth and Colocation Provided by http://
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
?
>
> Greetings
> Max
>
>
> - Nachricht von Leandro Dardini -
> Datum: Thu, 15 Sep 2016 18:06:14 +0200
>Von: Leandro Dardini
> Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>
I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.
An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium
As you know, there is the following settings
[general]
cachertclasses=yes ; use 1 instance of moh class for all users who are
using it,
; decrease consumable cpu cycles and memory
; disabled by default
It allows to use a single instance of MOH for all users
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
Hello,
I'd like to know if anyone of you is finding my same problems using any
recent asterisk version, after 13.7 / 13.8 with chan_sip.
If I use any recent asterisk version, after just few seconds asterisk
completely locks up, stopping processing SIP/UDP packets. Nothing is
written in the asteri
Hello,
I am moving from realtime chan_sip to pjsip and one of the problem I am
facing is the lack of "sipregs". With chan_sip, when an extension
registers, the server where it has registered to is stored in sipregs.
Is there something similar in pjsip? How can I find on which server the
pjsip exte
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 (Module res_odbc exactly) after the
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 was ambiguous.
>
> On We
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
I have some problem finding a smart way to add inbound trunks ip
authentication. I don't want to set allowguests=yes
Some of my providers just list some IP and I add them like:
[provider](!)
context=fromoutside
type=friend
insecure=port,invite
disallow=all
allow=g729
allow=ulaw
allow=alaw
canrein
Hello,
I just noticed a weird behavior when using ODBC functions. If the content
of any of the paramter has a "=" inside, then the function is not processed
correctly by asterisk.
Let's take for example the following ODBC function in func_odbc.conf
[LOG_SMS]
dsn=asterisk1,asterisk2
synopsis=Log t
Hello,I'd like to use a feature code for stopping recordings. Things are
quite easy when the call is received from the outside or just dialed from
inside to outside, but it can go really crazy when there are blind and
attended transfer going on. It ends I don't know on which call leg is the
recordi
Some time to time, usually after an asterisk restart or a sip reload, some
realtime sip peers are loaded in memory without their mailbox. I was not
able to replicate the issue on a constant basis, but after adding some
additional logs to asterisk, it seems the "add_peer_mailboxes" is run
correctly,
Hello,
I am facing a problem I can't understand. I have several realtime SIP peers
and from time to time, the mailbox field is not loaded in asterisk memory.
The mailbox field is correctly populated in the database, but often, after
an asterisk restart, the mailbox is not associated to the peer (ju
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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Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
under, like instead happen for example in the queue with the
setinterfacevar field.
I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.
Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is easy by using the dial
option U(...), but if I dial two extensions at once, when
Hello,
I am experiencing a weird problem on asterisk when I place an outbound
call, park it and then retrieve it. I am using extensions.ael with macro
and switch and I get something as SW_456_... that is autogenerated by
asterisk when compiling the extensions.ael
This doesn't happen when the call
Hello,
while most of the physical phones have keys to handle attended and blind
transfer, most soft phones have no support for it. Asterisk offers a
"featuremap" to assign a key to blindxfer and atxfer and they work fine if
the call is still in the same starting context, but if the call has moved
i
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
mand" command, and "sip show subscriptions" as a parameter
>
> --
>
> Alex Epshteyn
> email: a...@thirdlane.com
> web: www.thirdlane.com
> phone +1 415.261.6601
>
>
> - Original Message -
> > From: "Leandro Dardini"
> > To
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
--
___
Starting with asterisk 1.8, when you dial multiple channels at once and one
of them is answered, all other channels were canceled with the cause 200 -
Call completed elsewhere, so modern phones don't display the call as
"missed".
Do you know a way to transmit this cause over multiple channels? Let
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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New to Asterisk? Join us fo
Hello,
I am struggling to have a SPA504G to auto answer (for intercom/paging). I
have tried the following SIP headers (not all together), but without luck:
SIPAddHeader(Call-Info:\;answer-after=0);
SIPAddHeader(Call-Info: answer-after=0);
SIPAddHeader(Alert-Info: info=intercom);
SIPAddHeader(Alert
Hello,
while moving from asterisk 12.3 to asterisk 12.6, I see the MWI support for
voicemail has stopped working. If I check "sip show peer 104-DEVEL" on
asterisk 12.3, I can clearly see the "Mailbox" option set, while on
asterisk 12.6 it appears empty.
Is there anything to do more for having MWI
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
--
___
o the same file (using the a
> option)
>
>
> On 27 August 2014 21:20, Leandro Dardini wrote:
>
>> 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.
>&g
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 :
>
> Hello,
>
> After a sm
When a call is transferred to another extension using a blind transfer,
asterisk keeps traces of who is transferring in the BLINDTRANSFER variable.
If instead the call is "forwarded" using most phone call forward feature, a
302 Moved Temporarily is sent back to asterisk
-- Called SIP/104-DEVEL
About a call not being hang up for asterisk while the client hang up,
please remember SIP is based on UDP and UDP packets get easily lost... they
are retransmitted but sometime they are lost as the previous...
For the ghost calls, are the SIP port of the phones reachable from the
Internet... maybe
How long is the registration timeout? If the device is behind a
router/firewall, then you need to set a registration timeout lower than the
state table "life" in the router/firewall. I usually set my devices to just
2 minutes and it works almost all the time. Most Cisco devices have a very
long tim
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 functionality to wor
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
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, expensive bug still running in the code we are relaying
upon...
The problem is simple. If you transfer a call, that dialing will be no
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 mark 6
> == Using SIP RTP
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 issue was Asterisk pa
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:
> ---
>
> Current Sessions :
Is directmedia set to no?
>
>
> On 15 January 2014 23:11, Leandro Dardini wrote:
>
>> Hello,
>> I have an asterisk box with a peer configured with
>> nat=force_rport,comedia, but asterisk keeps sending the audio to the
>> private IP address and ignoring the cli
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the "sip show peer extension", I see both symmetric RTP and
Force Rport are set to yes, but asteri
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
Hello Mario,
nice to meet you on this mailing list!
Gigaset phones are a very high quality/price ratio, so I'll suggest you to
go with the dect ip models. Then you'll need to configure asterisk to act
as IVR, configure a queue and a failover to ring all hunt list.
Drop me a phone call and I'll be
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
> Hello Friends:
>
> I've just installed Asterisk 11 on my Linux (debian) server but it is not
> starting up when trying with "asterisk -vvc" and "service aster
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
Hello,
I was trying to use a queue in linear order and to provide the exact order
of members to dial by adjusting the uniqueid value. Obviously it doesn't
work and it seems an old problem:
https://issues.asterisk.org/jira/browse/ASTERISK-18480
Realtime configuration can't identify "orders" in the
Thu, Nov 14, 2013 at 3:54 AM, Leandro Dardini wrote:
>
>> Aligning presence over multiple servers is not simple and require some
>> changes on the dialplan and some custom code to transmit the state from one
>> server to the other.
>>
>> The BLF on the phone is d
Aligning presence over multiple servers is not simple and require some
changes on the dialplan and some custom code to transmit the state from one
server to the other.
The BLF on the phone is displayed using the "hint" of an extension. To be
able to manually manage the "hint" of an extension, you
2013/11/11 John T. Bittner
> Guys,
>
>
>
> I need you help on this one.
>
>
>
> Don’t know when this broke but we have a custom gui that runs on top of
> Asterisk running a real-time static for configurations.
>
> Nothing has changed with the database other than upgrades of Asterisk 10.
>
>
>
>
When you set sendrpid=yes in sip.conf, a very nice feature is activated.
When dialing an extension, the callerid of the dialed extension is returned
back on the display of the calling phone. So if you call extension 100, you
can see you are calling Ann (for example).
I want to selectively disable
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 Services
> which may be co
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, I send back.
> Philip
|
> | queue_lessthan | varchar(128) | YES |
> |||
> | queue_reporthold| varchar(128) | YES |
> |||
> | relative_periodic_announce | varchar(4)| YES | |
> 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
ha
scritto:
> - Original Message -
> > From: "Leandro Dardini"
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> > Sent: Tuesday, March 26, 2013 5:28:22 AM
> > Subject: [asterisk-users] rtcache
Hello friends,
I am using from a long time rtcachefirends=yes and rtautoclear=yes in
my sip.conf for asterisk 11.2.1.
I have found the data of the peers are never reloaded from the
database, so if you change the password for a peer, it will continue
to work with the old password. Do you think it i
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
2013/3/21 Florian Wolters :
> Hi @ll,
>
> I just moved my Asterisk Box and changed the Provider and Internet Access to
> a full IP Access by Deutsche Telekom.
>
> I set up my sip.conf as I found various examples throughout the Net. Calls
> and some other stuff is basically working.
>
> The proble
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 been using A
Top replying ...
In the CDR you have two fields, "duration" and "billed". "Duration" is
the total time from "Dial" command to end of calls. It is the time the
"Dial" command is running. "Billed" is the time from when the other
party answered and the end of the call.
In your example, duration and
2013/3/12 kepin sinatra
> hi, anybody know how to make a sip trunk between trixbox and panasonic kx
> tde 100?
> i've tried, but always failed when calling from trixbox to panasonic.
>
> thanks,,,
>
>
> --
>
Try a SIP debug to understand the reason it fails. Is it a problem of
codec? Is it a pro
2013/3/8 nik600
> Dear all
>
> i'm planning a migration to asterisk for a high volume IVR service
> (from 1000 to 1500 concurrent call)
>
> The IVR service is based only on DTMF tones so the features required is
>
> - play feature
> - dtmf detection
>
> Asterisk will receive calls via VOIP (SIP w
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
2013/3/7 Duncan Turnbull
>
> On 7/03/2013, at 9:29 PM, Kamlesh Kumar wrote:
>
>
> On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
>
> You can use ATA box with pstn phone to reduce cost.
>
>
> Are you wiring a building where multiple-line SIP gateways make sense?
>
> How about a description of what yo
2013/3/7 Steve Edwards
> Please don't top-post.
>
>
> On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
>
> You can use ATA box with pstn phone to reduce cost.
>>
>
> Are you wiring a building where multiple-line SIP gateways make sense?
>
> How about a description of what you are trying to do?
>
> Per
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, but I hav
Hi,
>
> You might want to use ${MACRO_EXTEN} variable inside to preserve exten
> variable of the original dialplan exten variable.
>
> Mitul
> On Feb 24, 2013 4:04 PM, "Leandro Dardini" wrote:
>
>> I just discover an "hidden" problem with AEL macro I wa
I just discover an "hidden" problem with AEL macro I want to have your
feedback. If you use a macro to dial out, like &dialout(${EXTEN}), the leg
extension will became s and if it happens you transfer the call,
that will be the callerid appearing on the other phone display.
I am just rewrit
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?
> My goal is to inject my di
2013/2/21 Enrico Pasqualotto
> Hi all, I'm trying to setup a Quiz/feedback for caller of call center when
> a agent hangup.
> I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c
> and g but every time I try to play something I got:
>
> -- Executing [301@from-test:1] Dial("
2013/2/21 akhilesh chand
> hello all,
>
> i have two asterisk server for call transfer and one more asterisk server
> for agent login(server_X) where agent take the call.
>
> server_A and server_B
> server_A is connected with pri and configure with 60 channel for call
> transfer into server_X
>
2013/2/20 Nguyễn Công
> Hello everyone, I’m new to Asterisk and I have a question. There is a
> phone call between two users, then they are talking to each other directly
> or by the server. I mean all packets from the user A to user B will be send
> directly to each other or will those packets f
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
2013/1/31 Ishfaq Malik
> On Wed, 2013-01-30 at 14:10 -0700, Carlos Alvarez wrote:
> > On Wed, Jan 30, 2013 at 12:05 PM, XBrian wrote:
> > Thanks - I was hoping there was some silver bullet to use out
> > there. Thanks
> > anyway.
> >
> >
> > There is. If you build a reli
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. If a so
2013/1/30 XBrian
> I am pulling my hairs out here. This is my dialplan.
>
> exten => 100,1,Set(AGISIGHUP=no)
> exten => 100,n,AGI(a2billing.php,4,callingcard)
> exten => 100,n,Set(__APP_MSG_IND=${APP_MSG_IND})
> exten => 100,n,Set(__APP_MESSAGE=${APP_MESSAGE})
> exten => 100,n,Hangup()
>
> exten
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
2013/1/26 RSCL Mumbai
> Hello,
>
> I have Elastix ISO install (FreePBX 2.7.0.3)
>
> My current Setup is as follows:
> Inbound Route > Queue > (Dynamic Agents)
>
> The queue distributes calls based on rrMemory.
>
> I have been asked to redesign the call distribution as follows:
>
> Calls will be d
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 unbelievable ... I wa
2013/1/25 Dan Journo
> >> Upgrading to the latest version didn't help. After about 30 minutes,
> Asterisk2 tries to send out OPTIONS keepalive packets to peers listed as
> Registered on Asterisk1.
>
> > It is something really amazing... Can you run "sip show peers" on each
> one of the server
2013/1/24 Dan Journo
> > I am curious, is your version of asterisk correctly compiling the
> regserver field? Each server needs to have a distinct server name.
>
> ** **
>
> Upgrading to the latest version didn't help. After about 30 minutes,
> Asterisk2 tries to send out OPTIONS keepalive pa
2013/1/24 Dan Journo
> >> 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?
>
> ** **
>
> ** **
>
> 1.8.15.0. I'll upgrade it to 1.8.20.1 when I can an
2013/1/23 Dan Journo
> > Maybe you are lacking some of the configuration. These is the relevant
> part.
>
> ** **
>
> > rtcachefriends=yes
>
> > rtsavesysname=yes
>
> > rtupdate=yes
>
> > rtautoclear=yes
>
> ** **
>
> We have
>
> rtcachefriends=yes
>
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
2013/1/23 Dan Journo
> Hi,
>
> ** **
>
> We're trying to decide whether to switch back to a static file for
> sip.conf. Currently we use mysql realtime but can't see any real benefit.*
> ***
>
> ** **
>
> Why would someone choose realtime sip over static files?
>
> ** **
>
> Thanks
>
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
> dialplan when usin
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