Displaying 20 results from an estimated 32 matches for "saraiva".
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saraniva
2018 Nov 30
2
Asterisk non-root - selinux - astdb
...: SQL logic
error or missing database
CentOS 7.5.1804
Asterisk certified/13.21-cert3
[root at sv03 asterisk]# ls -lahZ /var/lib/asterisk/astdb.sqlite3
-rw-r-----. asterisk asterisk unconfined_u:object_r:asterisk_var_lib_t:s0
/var/lib/asterisk/astdb.sqlite3
Can anyone help?
Thanks.
Rafael S. Saraiva
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2013 May 18
5
Performance Asterisk large installation on Vmware/Xen
...400 extensions. My question is whether this scenario carry an Asterisk
virtualized. Will be used only extensions and trunks sip sip, 1 queue with
2 agents, without call recording. It is best to use XEN or VMware? Which
best version of Asterisk for this scenario?
Thank you.
Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956 | (51) 3205-1504
http://www.astdocs.com | <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2015 May 12
1
AEL keyword IfTime with variable on time range
...rocess: parsed config file name '/etc/asterisk/extensions.ael'.
[May 12 14:31:52] ERROR[20773]: pbx_ael.c:197 pbx_load_module: Sorry, but 1
syntax errors and 0 semantic errors were detected. It doesn't make sense to
compile.
[image: Sua Foto] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS
| Mobile: (51) 8174-7956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
<https://plus.google.com/u/0/+RafaelSaraivaRS>
2015-05-12 13:51 GMT-03:00 Tech Support <asterisk at voipbusiness.us>:
> You should try it and find out if it works. If it does, le...
2017 Oct 06
2
Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP
Hi
Is it a normal behavior of Asterisk put a call on hold when receive a
Session Progress with media address 0.0.0.0 in SDP? I believe the call on
hold should be initiate with a re-invite.
Thanks
--
Att,
Rafael Saraiva
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2015 May 12
2
AEL keyword IfTime with variable on time range
Hi
It's possible using a variable in the iftime keyword argument?
E.g:
context text {
s => {
timerange = '06:00-12:00|*|*|*';
ifTime(${timerange} {
Playback(ivr/goodbye);
}
}
}
thanks
[image: Sua Foto] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS
| Mobile: (51) 8174-7956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
<https://plus.google.com/u/0/+RafaelSaraivaRS>
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2019 Feb 15
2
Set qualify = yes on trunk can't do outgoing call
Hello when I set qualify = yes on trunk I can't do outgoing call.
Incoming is always working.
[Feb 15 23:01:41] WARNING[12909][C-00000012]: app_dial.c:2525
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
but my linphone is registered all the time.
when set qualify = no outgoing call is working
(but i have problems when WAN IP is changed after
2014 Mar 26
1
Verbose only one context
Hi
It's possible in Asterisk 1.8 enable verbose only in one context or
extension?
thanks
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2014 Mar 31
1
Function REGEX
...if is possible test a
expression as [X123 == 5123] ( If an extension corresponding to a
previously defined regular expression). I saw various examples about this
function, but nothing as the my needs. I do not understanding exactly how
to works this function.
Thank's
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2013 Aug 05
3
Voicemail variables on email subject
...used
voicemail variables, e.g. :
voicemail.conf
emailsubject=${VM_MAILBOX}|${VM_MSGNUM}|${VM_CALLERID}|${VM_DUR}
Return:
Subject: =?utf-8?Q?1504|12|=22Teste_-_Rafael=22_=3C1570=3E|0=3A16?=
Expected:
Subject: 1504|12|"Teste - Rafael" <1570>|16
Thank's
Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2014 Jun 30
2
Sippeers realtime with minimum table
Hi there
It's possible configure realtime mysql in Asterisk with a non standard
sippeers table?
I need using a sippeers table from other system (non Asterisk). This table
has a minimal configuration.
Thank's
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2014 Jul 11
2
CDR(dst) not set in AEL macro
...estno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} > 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}
Thank's.
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2015 Aug 12
2
Busy level in Asterisk 11
...bscribe=yes
[100](template)
type=friend
context=default
host=dynamic
secret=***
[101](template)
type=friend
context=default
host=dynamic
secret=***
[102](template)
type=friend
context=default
host=dynamic
secret=***
Thanks in advance
[image: Sua Foto] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS
| Mobile: (51) 8174-7956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
<https://plus.google.com/u/0/+RafaelSaraivaRS>
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2015 May 12
0
AEL keyword IfTime with variable on time range
You should try it and find out if it works. If it does, let us know.
Regards;
John
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Rafael dos Santos Saraiva
Sent: Tuesday, May 12, 2015 11:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] AEL keyword IfTime with variable on time range
Hi
It's possible using a variable in the iftime keyword argument?
E.g:
context text {
s => {
timerang...
2011 May 19
1
Pridialplan/ prilocaldialplan
...diaplan in chan_dahdi.conf. I interconnect the Asterisk with a
Siemens PBX, but i saw that the changes in the file do not take effect in
debug of the span or calling/called number. How to use this options? In that
cases to use?
Ps.: sorry for the english, i'm brazilian.
Thanks
--
Att,
Rafael Saraiva
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2017 Feb 17
6
Turn on SIP debugging from DialPlan
I have some troublesome numbers that I would like to capture the SIP
dialogue when I am calling them. When I am about to dial the number, is
there any way to turn on SIP debugging in the dial plan before I make the
call? (and turn it off after the call is completed?)
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2013 Aug 19
0
Reverse Charging Indication <> MFCR2
Hi
It's possible verify the Reverse Charging Indication on mfcr2 link directly
con dialplan?
Thank's
Att,
*Rafael dos Santos Saraiva*
Tel: (51) 8174-7956
*Digium Certified Asterisk Administrator (dCCA)*
http://www.astdocs.com | <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2014 Apr 03
1
func_odbc
Hi All
Anyone know how to do include files with func_odbc.conf?
I now have several pages of functions in my func_odbc.conf and it is
getting harder to maintain it.
I would like to break them up into files by category. The standard method
of using the #include does not seem to work .
Ideas are appreciated.
Bryant
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2014 Jul 10
0
CDR(dst) in AEL macro
...(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} > 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}
Thank's.
Att,
*Rafael dos Santos Saraiva*
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
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2014 Jul 24
1
audio gain in SIP channel
hello all,
i'm trying to do what in object with an asterisk box 11.11 on centos6.5,
using functions
AGC and VOLUME, but seems that does not work at all.
There is a way to check this values during setup/call?
Maybe is it not possible realize what i'd like to do?
Could anyone can help me on this?
thanks a lot in advance
regards
Lorenzo
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2014 Oct 28
2
Asterisk 13 stable?
Hi
The Asterisk 13 is already stable for production environment?
thank's
[image: Sua Foto] <rafaelsnsa at gmail.com>Rafael S. SaraivaPorto Alegre - RS
| Mobile: (51) 8174-7956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
<https://plus.google.com/u/0/+RafaelSaraivaRS>
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