Re: [asterisk-users] Set qualify = yes on trunk can't do outgoing call

[Rafael dos Santos Saraiva]

You can manipulate the options defaultexpiry and maxexpiry in global
section, so that decrease the interval betwen the REGISTER messages and
minigate the problem of changing wan ip.



Rafael S. Saraiva
Porto Alegre - RS | Mobile: (51) 981-747-956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>



Em sex, 15 de fev de 2019 às 20:40, basti 
escreveu:

> Hello, asterisk think my local phone (extension 20) is absent.
>
> On 15.02.19 23:26, Rafael dos Santos Saraiva wrote:
> > Hi
> >
> > When you set qualify to yes, the asterisk "test" the sip trunk with
> > OPTIONS messages, if no receive responses from this messages, it
> > consider the trunk offline. Possibly your sip provider dont accept (and
> > dont reply) sip options requests.
> >
> >
> >   Rafael S. Saraiva
> > Porto Alegre - RS | Mobile: (51) 981-747-956
> > [View Rafael Saraiva's profile on LinkedIn]
> > <http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>
> >
> >
> >
> > Em sex, 15 de fev de 2019 às 20:14, basti  > <mailto:mailingl...@unix-solution.de>> escreveu:
> >
> > 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-0012]: 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 reconnect internet
> > connection)
> >
> > how can i solve this?
> > best regards
> >
> > --
> > _
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> > https://community.asterisk.org/
> >
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
>
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Re: [asterisk-users] Set qualify = yes on trunk can't do outgoing call

[Rafael dos Santos Saraiva]

Hi

When you set qualify to yes, the asterisk "test" the sip trunk with OPTIONS
messages, if no receive responses from this messages, it consider the trunk
offline. Possibly your sip provider dont accept (and dont reply) sip
options requests.


Rafael S. Saraiva
Porto Alegre - RS | Mobile: (51) 981-747-956




Em sex, 15 de fev de 2019 às 20:14, basti 
escreveu:

> 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-0012]: 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 reconnect internet
> connection)
>
> how can i solve this?
> best regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Asterisk non-root - selinux - astdb

[Rafael dos Santos Saraiva]

Hi Jean

Thanks, you've solved my problem.

Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=1342733


Solution:

semanage fcontext -a -t asterisk_var_lib_t /var/lib/asterisk/

restorecon -v /var/lib/asterisk/

Regards

Rafael S. Saraiva
Porto Alegre - RS | Mobile: (51) 981-747-956
<http://br.linkedin.com/pub/rafael-saraiva/52/aab/230>



Em seg, 3 de dez de 2018 às 05:51, Jean Aunis 
escreveu:

> Hello,
>
> I haven't tried but this post probably gives a solution :
>
> https://bugzilla.redhat.com/show_bug.cgi?id=1342733
>
> Regards
>
> Jean Aunis
> Le 30/11/2018 à 19:24, Rafael dos Santos Saraiva a écrit :
>
> Hi
>
> I'm trying to use Asterisk running as non-root user and selinux enabled.
> Asterisk is running ok, but astdb not works. When i try to put in astdb,
> console shows this message:
>
> WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic
> error or missing database
>
> CentOS 7.5.1804
> Asterisk certified/13.21-cert3
>
> [root@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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>
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>
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[asterisk-users] Asterisk non-root - selinux - astdb

[Rafael dos Santos Saraiva]

Hi

I'm trying to use Asterisk running as non-root user and selinux enabled.
Asterisk is running ok, but astdb not works. When i try to put in astdb,
console shows this message:

WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic
error or missing database

CentOS 7.5.1804
Asterisk certified/13.21-cert3

[root@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

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Re: [asterisk-users] Reject call from Asterisk dialplan

[Rafael dos Santos Saraiva]

Hi

I guess is not possible send specific SIP response from dialplan in
Asterisk, but you can send ISDN hangupcauses. In this case, to reject the
call you can use Hangup(21).
To do this remotly, my suggestion is create a context that pickup the call
and execute hangup with cause 21.


[image: Sua Foto]  Rafael S. Saraiva
Porto Alegre - RS | Mobile: (51) 981-747-956



2018-05-08 15:12 GMT-03:00 Alexander Lopez :

> Use a script to redirect the ringing call into an extension that returns
> the proper sip result, and hangup.
>
>
>
> You could also add logic to alert or log that call.
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] *On Behalf Of *Mike
> *Sent:* Tuesday, May 08, 2018 1:46 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> asterisk-users@lists.digium.com>
> *Subject:* [asterisk-users] Reject call from Asterisk dialplan
>
>
>
> Hi,
>
>
>
> I’m looking for a way to reject a call remotely using the Asterisk
> dialplan.
>
>
>
> For example, phone A is ringing – I’m at the other end of the room next to
> phone B, and I want to reject the call to Phone A by dialing an extension.
>
>
>
> I’m basically trying to reproduce the Polycom “reject” action but through
> the Asterisk dialplan.
>
>
>
> Reasons:
>
>1. It would allow me to log through Asterisk who’s rejecting calls
>2. It would allow rejecting calls from another phone  (see above
>scenario)
>
>
>
> I thought there could be a “SendSIPCode 486 to SIP peer phoneA”
> application, but a quick scan of the documentation does not bring obvious
> answers.
>
>
>
> Mike
>
>
>
>
>
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> org/
>
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Re: [asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

[Rafael dos Santos Saraiva]

The problem the 183 is received with mode sendonly and playbacks an audio,
so when the destination playbacks audio, the origin was put on hold.

 |   |
 |  INVITE   |
 |-->|
 | 183 Session Progress/SDP  |
 |<--|
 |  RTP one way  |
 |<--|
 |BYE|
 |<--|
 |   |


This method is used by telco to playback message to not found user.




2017-10-06 3:25 GMT-03:00 Jean Aunis <jean.au...@prescom.fr>:

> I think it is normal, the call is placed on hold as soon as the remote
> media address is null.
>
> It makes sense because when a 183 is sent, some media is supposed to be
> sent as with a 200, so placing the call on hold when no media is available
> sounds logic.
>
> Le 06/10/2017 à 03:56, Rafael dos Santos Saraiva a écrit :
>
> 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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> org/
>
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>
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>



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[asterisk-users] Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP

[Rafael dos Santos Saraiva]

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

-- 
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Re: [asterisk-users] Turn on SIP debugging from DialPlan

[Rafael dos Santos Saraiva]

Hi

I don't know if works, but you can try this:

System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
or udp portrange 1-2 &);
Wait(1);
Dial(SIP/${EXTEN});
System(pkill tcpdump);
Hangup;

Or whitout RTP:

System(tcpdump -nq -s 0 -i eth0 -w /tmp/sip.pcap port 5060
&);
Wait(1);
Dial(SIP/${EXTEN});
System(pkill tcpdump);
Hangup;

Probably the last messages of SIP will be lost, BYE for example.





2017-02-17 20:43 GMT-02:00 Derek Andrew :

> 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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[asterisk-users] PRI error: link goes down when make calls

[Rafael dos Santos Saraiva]

Hi

I have a PRI link with a Brazilian telco when i make a call from Asterisk
to PRI link the call doesn't complete and the link goes down (RED alarm),
after this returns to status OK. Incoming calls works, but whitout audio

The following link has the log on call at the moment when the link becomes
down:

http://pastebin.com/V5ySVc0f

Any suggestion?

Thank's


-- 

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[asterisk-users] DAHDI Dynamic span as INT device identificator

[Rafael dos Santos Saraiva]

Hi

Is it possible assign an INT identification to dynamic device in DAHDI?
Example:

This is device: DYN/eth/eth1/04:74:A1:00:0A:AE/0
I want call this as span 1

I saw the assigned-spans.conf and aparently can be this.

Thanks in advance.


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Re: [asterisk-users] using dynamic DAHDI loop back

[Rafael dos Santos Saraiva]

Hi
Insert this on first line of chan_dahdi.conf:
[channels]



[image: Sua Foto]  Rafael S. Saraiva
Porto Alegre - RS | Mobile:  (51) 8174-7956



2016-03-13 10:01 GMT-03:00 Mehdi Shirazi :

> 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
> echocancelwhenbridged=no
> context=from-pstn
> switchtype = euroisdn
> signalling = pri_net
> channel => 1-15,17-31
> group=1
> switchtype = euroisdn
> signalling = pri_cpe
> channel => 32-46,48-62
>
> all dahdi command line outputs like dahdi_tool are ok but inside Asterisk
> there is
> nothing in "Dahdi show channels" .
> please help me
>
> Regards
>
> --
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[asterisk-users] set framing on dynamic interface DAHDI

[Rafael dos Santos Saraiva]

Hi

I working with DAHDI Dynamic Interfaces using ethernet boards. I need set
the framing to CCS, but the documentation of DAHDI not refer to it. My
question is: there is a way to do this?

*system.conf*
dynamic=eth,enp0s8/00:00:00:00:00:01/0,31,0
echocanceller=mg2,1-15,17-31
bchan=1-15,17-31
dchan=16
alaw=1-15,17-31

*dahdi_scan*
[1]
active=yes
alarms=RED
description=Dynamic 'eth' span at 'enp0s8/00:00:00:
name=DYN/eth/enp0s8/00:0
manufacturer=
devicetype=DYN/eth/enp0s8/00:00:00:00:00:01/0
location=
basechan=1
totchans=31
irq=0
type=digital-DYNAM
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI,HDB3
framing_opts=ESF,D4,CCS,CRC4
coding=
framing=CAS


Thank's in advance.



[image: Sua Foto]  Rafael S. Saraiva
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Re: [asterisk-users] Manipulating of a dialed sequence

[Rafael dos Santos Saraiva]

Hi

Try this:

[pbx]
exten => _1X.,1,Answer()
same  => n,Set(VAR=012345*543210)
same  => n,Set(VAR1=${CUT(VAR,*,1)});012345
same  => n,Set(VAR1=${CUT(VAR,*,2)});543210



[image: Sua Foto] Rafael S. SaraivaPorto Alegre - RS
| Mobile:  (51) 8174-7956



2015-12-05 14:21 GMT-02:00 Frank :

> Hi Asterisk List
>
> Given, as an example, the following sequence
>
> 012345*543210
>
> I would like to store into a variable all digits before "*" (012345) and
> in a different variable all digits after the "*" (543210) for further
> processing in the dial plan.
>
> The length of the dialed sequence may be variable and "*" is the
> separator between the two values to store.
>
> Any idea?
>
> Thanks
>
> Francesco
>
>
>
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Re: [asterisk-users] Busy level in Asterisk 11

[Rafael dos Santos Saraiva]

Hi Joshua

Thanks for you explanation.

I found another problem. When I reject call by group limit, the CCSS do not
work, only if call limit on sip peers is set to 1. There is a way to using
ccss when call limit is different to one?

Thanks in advance.


[image: Sua Foto] rafaels...@gmail.comRafael 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

2015-08-12 9:41 GMT-03:00 Joshua Colp jc...@digium.com:

 On Wed, Aug 12, 2015, at 09:34 AM, Rafael dos Santos Saraiva wrote:
  Hi

 Kia ora,

  I need to set the number  of incoming calls to one, but the outgoing
  calls
  should be unlimited. I think the busylevel parameter is for it(incoming
  calls), but not works. My config is:

 snip

 The busylevel configuration option controls device state:

 ; If you set the busylevel, we will indicate busy when we have a number
 of calls that
 ; matches the busylevel treshold.

 That is, if the number of calls is equal to the busylevel then the state
 of the device is considered busy. It does not prevent calls from
 occurring.

 The call-limit configuration option is what prevents the calls, and it
 has no separation between inbound and outbound. You would have to use
 other constructs (such as group counting in the dialplan) to do it.

 Cheers,

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Busy level in Asterisk 11

[Rafael dos Santos Saraiva]

Hi

I need to set the number  of incoming calls to one, but the outgoing calls
should be unlimited. I think the busylevel parameter is for it(incoming
calls), but not works. My config is:

cat sip.conf
[general]

[template](!)
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
call-limit=2
busylevel=1
callcounter=yes
subscribecontext = hint
allowsubscribe=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] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS
| Mobile:  (51) 8174-7956
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
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[asterisk-users] Update of dialed number on sip phones

[Rafael dos Santos Saraiva]

Hi

I have a dialplan that search a phone from dialed code, i.e:

mysql table:
code:1234
dest: +555133449966

query in odbc function:
SELECT dest FROM my_table WHERE code = '${ARG1}'

dialplan:
exten = _#7,1,Set(DESTNO=${ODBC_query_dest_in_table(${EXTEN:2})})
same  = n,Dial(SIP/xxx.xxx.xxx/${DESTNO},30,tT)
same  = n,hangup

I need to update the dialed number in the screen of SIP phones, when query
search the number, the Asterisk should update the #7 for the found
number. I know that this facility probably should to be supported by the
phones. I tried change the parameters like pid, rpid, but no success. In
Asterisk, is it possible? Which parameters need I change? And in the sip
phone, which facilities should it have?

Thanks in advance.



[image: Sua Foto] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS
| Mobile:  (51) 8174-7956
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
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Re: [asterisk-users] Calling multiple phones at ones

[Rafael dos Santos Saraiva]

Hi Ivan

Using the following extensions:
SIP/100
SIP/101
SIP/102

Example 1:
[default]
exten = _X.,1,Dial(SIP/100SIP/101SIP/102,30,tT)
same = n,hangup

Example 2:
exten = _X.,1,Dial(SIP/100,10,tT)
exten = _X.,2,Dial(SIP/101,10,tT)
exten = _X.,3,Dial(SIP/102,10,tT)
same = n,hangup

This is the simplest methods, but there are others more complex and better.


[image: Sua Foto] rafaels...@gmail.comRafael 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

2015-06-14 23:12 GMT-03:00 Ivan Demkovitch idemkovi...@yahoo.com:

 Hello group!

 I’m new to Asterisk but got one running finally :)

 Now I’m trying to solve following problem. I have company Automated
 Attendant and each employee have
 SIP phone at home, SIP phone in office, cell phone.

 I want all those 3 phones to be “one”. So, if someone calls our company
 number and dials my extension - I’d like 3 phones to ring at the same time.

 What is this feature and where should I look for samples, etc? I’m going
 by “Asterisk: The definite guide” book and pretty confident with those
 concepts described but not sure
 how to achieve what I described above.

 Thank you,
 Ivan
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Re: [asterisk-users] AEL keyword IfTime with variable on time range

[Rafael dos Santos Saraiva]

Solved!!


Set(time_01=06:00-12:00,*,*,*);
Set(time_02=12:00-18:00,*,*,*);
Set(time_03=18:00-06:00,*,*,*);

if(${IFTIME(${time_01}?1:0)} == 1) {
NoOp(Bom dia);
Playback(beep);
} else if(${IFTIME(${time_02}?1:0)} == 1) {
NoOp(Boa tarde);
Playback(beep);
} else if(${IFTIME(${time_03}?1:0)} == 1) {
NoOp(Boa noite);
Playback(beep);
}


Thank's



[image: Sua Foto] rafaels...@gmail.comRafael 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 14:39 GMT-03:00 Rafael dos Santos Saraiva rafaels...@gmail.com:

 Sorry, I forget to tell I tried, but not works.

 *Context:*
 context ivr_temp2 {
 s = {
 Proceeding();
 str_time_01 = '06:00-12:00|*|*|*';  // Manhã
 ifTime (${str_time_01}) {
 Playback(ura/bom_dia);
 }
 }
 }

 The error is showed on ael reload.

 *Console errors:*
 rssr304*CLI ael reload
 Command 'ael reload' failed.
 [May 12 14:31:52] NOTICE[20773]: pbx_ael.c:164 pbx_load_module: Starting
 AEL load process.
 [May 12 14:31:52] ERROR[20773]: ael.y:840 ael_yyerror:  File:
 /etc/asterisk/extensions.ael, Line 315, Cols: 32-32: Error: syntax error,
 unexpected ')', expecting '|'
 [May 12 14:31:52] NOTICE[20773]: pbx_ael.c:177 pbx_load_module: AEL load
 process: 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] rafaels...@gmail.comRafael 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 aster...@voipbusiness.us:

 You should try it and find out if it works. If it does, let us know.

 Regards;

 John



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@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 = {

   timerange = '06:00-12:00|*|*|*';

   ifTime(${timerange} {

Playback(ivr/goodbye);

   }

  }

 }





 thanks




 [image: Image removed by sender. Sua Foto] rafaels...@gmail.com

 *Rafael S. Saraiva*

 Porto Alegre - RS | Mobile: [image: Image removed by sender.] (51)
 8174-7956

 [image: Image removed by sender.]
 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 [image: Image
 removed by sender.] https://plus.google.com/u/0/+RafaelSaraivaRS



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Re: [asterisk-users] AEL keyword IfTime with variable on time range

[Rafael dos Santos Saraiva]

Sorry, I forget to tell I tried, but not works.

*Context:*
context ivr_temp2 {
s = {
Proceeding();
str_time_01 = '06:00-12:00|*|*|*';  // Manhã
ifTime (${str_time_01}) {
Playback(ura/bom_dia);
}
}
}

The error is showed on ael reload.

*Console errors:*
rssr304*CLI ael reload
Command 'ael reload' failed.
[May 12 14:31:52] NOTICE[20773]: pbx_ael.c:164 pbx_load_module: Starting
AEL load process.
[May 12 14:31:52] ERROR[20773]: ael.y:840 ael_yyerror:  File:
/etc/asterisk/extensions.ael, Line 315, Cols: 32-32: Error: syntax error,
unexpected ')', expecting '|'
[May 12 14:31:52] NOTICE[20773]: pbx_ael.c:177 pbx_load_module: AEL load
process: 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] rafaels...@gmail.comRafael 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 aster...@voipbusiness.us:

 You should try it and find out if it works. If it does, let us know.

 Regards;

 John



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@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 = {

   timerange = '06:00-12:00|*|*|*';

   ifTime(${timerange} {

Playback(ivr/goodbye);

   }

  }

 }





 thanks




 [image: Image removed by sender. Sua Foto] rafaels...@gmail.com

 *Rafael S. Saraiva*

 Porto Alegre - RS | Mobile: [image: Image removed by sender.] (51)
 8174-7956

 [image: Image removed by sender.]
 http://br.linkedin.com/pub/rafael-saraiva/52/aab/230 [image: Image
 removed by sender.] https://plus.google.com/u/0/+RafaelSaraivaRS



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[asterisk-users] AEL keyword IfTime with variable on time range

[Rafael dos Santos Saraiva]

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] rafaels...@gmail.comRafael 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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[asterisk-users] Asterisk 13 stable?

[Rafael dos Santos Saraiva]

Hi

The Asterisk 13 is already stable for production environment?

thank's





[image: Sua Foto] rafaels...@gmail.comRafael S. SaraivaPorto Alegre - RS
| Mobile:  (51) 8174-7956
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
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Re: [asterisk-users] audio gain in SIP channel

[Rafael dos Santos Saraiva]

Hi

To using VOLUME function the syntax is:
Set(VOLUME(rx)=+n)
Set(VOLUME(rx)=-n)
Set(VOLUME(tx)=+n)
Set(VOLUME(tx)=-n)

I think is not possible retrieve the value of the channel.




Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-24 7:52 GMT-03:00 lore pino.mai...@gmail.com:

 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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Re: [asterisk-users] audio gain in SIP channel

[Rafael dos Santos Saraiva]

I dont using these functions (AGC/ DENOISE). My suggestion... try invert
the priorities:
Set(DENOISE(tx)=on)
Set(DENOISE(rx)=on)
Set(AGC(rx)=)
Set(AGC(rx)=)

And try higher values.. is more easy the perception if the values are
larger than default.


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-24 9:23 GMT-03:00 lore pino.mai...@gmail.com:

 thanks a lot Rafael.
 could you tell me also something about AGC(rx)=?
 I mean, i've tryed

 Set(AGC(rx)=)
 Set(AGC(rx)=)
 Set(DENOISE(tx)=on)
 Set(DENOISE(rx)=on)

 using =8000, 16000 and 32000 but all calls looked like to have se same
 audio gain.

 thanks for your rapid reply.


 2014-07-24 13:12 GMT+02:00 Rafael dos Santos Saraiva rafaels...@gmail.com
 :

 Hi

 To using VOLUME function the syntax is:
 Set(VOLUME(rx)=+n)
 Set(VOLUME(rx)=-n)
 Set(VOLUME(tx)=+n)
 Set(VOLUME(tx)=-n)

 I think is not possible retrieve the value of the channel.




 Att,
 *Rafael dos Santos Saraiva*
  http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


 2014-07-24 7:52 GMT-03:00 lore pino.mai...@gmail.com:

  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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 --
 Chi vive sperando muore cagando ... Lo Russo isoletta dell'Egeo che non
 conta un cazzo, 1941 ... sono anche un autore

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Re: [asterisk-users] Call Identifier Logging

[Rafael dos Santos Saraiva]

Hi
I tried this in ael:
_000. = {
Proceeding();
callident = ${SHELL(asterisk -rx core show channel
${CHANNEL} | grep Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d]
-f1 | cut -d\n -f1):0:-1};
NoOp(${callident}});
Dial(Motif/google/+${EXTEN:3}@voice.google.com,,r);
hangup;
}

And worked perfectly.

It would be interesting, the developer team add a variable to channel with
this data.




Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-21 18:59 GMT-03:00 Steven Wheeler swhee...@usinternet.com:

  Hello,

 I am working on upgrading from Asterisk 1.8 to Asterisk 11.6. One of the
 features we are excited for is Call Identifier Logging
 https://wiki.asterisk.org/wiki/display/AST/Call+Identifier+Logging.
 However, it doesn't appear that this new Call ID is accessible from the
 dial plan. Ideally we would like to store this Call ID in the CDR. Does
 anyone know if this is possible?



 I could do something like this, but it seems like a terrible hack:

 same = n,Set(CALLID=${SHELL(asterisk -rx core show channel ${CHANNEL} |
 grep ' Call Identifer' | egrep -o 'C-[0-9a-f]+')})



 Also as a side note, in the core show channel output ' Identifier' is
 misspelt as ' Identifer'

 *Steven Wheeler*



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Re: [asterisk-users] Call Identifier Logging

[Rafael dos Santos Saraiva]

Try this:
CDR(userfield) = ${SHELL(asterisk -rx core show channel ${CHANNEL} |
grep Call Identifer | cut -d: -f2 | cut -d[ -f2 | cut -d] -f1 | cut -d\n
-f1):0:-1};


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-22 15:08 GMT-03:00 Steven Wheeler swhee...@usinternet.com:

  Making LinkedID available in the dialplan would also be useful.

 LinkedID is already available in the dialplan: CHANNEL(linkedid)

 Which version was that added?  I don’t see it on my 11.10.0



 [daffy-01 ~]# asterisk -rx core show function CHANNEL | grep -i link

 [daffy-01 ~]#





 According to funcs/func_channel.c

 468 else if (!strcasecmp(data, linkedid)) {

 469 ast_channel_lock(chan);

 470 if (ast_strlen_zero(ast_channel_linkedid(chan))) {

 471 /* fall back on the channel's uniqueid if
 linkedid is unset */

 472 ast_copy_string(buf,
 ast_channel_uniqueid(chan), len);

 473 }

 474 else {

 475 ast_copy_string(buf,
 ast_channel_linkedid(chan), len);

 476 }

 477 ast_channel_unlock(chan);



 While useful, that doesn't solve the problem of being able to store the
 channel's logging identifier in CDR.



 *Steven Wheeler*

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Re: [asterisk-users] Call Identifier Logging

[Rafael dos Santos Saraiva]

Really, a dialplan function would be best. I too don't like of an idea of
using a external process to get internal variables, but when necessary...
 :(


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-22 16:29 GMT-03:00 Steve Edwards asterisk@sedwards.com:

 On Tue, 22 Jul 2014, Steve Edwards wrote:

  How about something like:

 asterisk -rx core show channel SIP/spa841-0003\
 | awk '/Call Identifer/ {gsub(/[][]/,); print $3}'


 Or:

 asterisk -rx core show channel SIP/spa841-0003\
 | awk -F'[][]' '/Call Identifer/ {print $2}'

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] CDR(dst) not set in AEL macro

[Rafael dos Santos Saraiva]

Hi

Probably are this bug, my Asterisk version is 1.8.15.0(the version of
report is 1.8.15.1).

I see that the problem occurs when the call is answered, if busy, fail,
unanswered, the field dst is correct. I see too that when the problem
occurs the dcontext field is set as name of the macro(aparently, the return
in macro no works).

 I will go try with others versions and report the status.

Thank's.


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-12 7:33 GMT-03:00 Johan Wilfer li...@jttech.se:

 2014-07-11 15:38, Rafael dos Santos Saraiva skrev:

 Hi

 I'm using a macro to dial in a AEL dialplan. The problem is the macro do
 not set the field  CDR(dst), showing only ~~s~~.

 I tried various configurations, but without solutions.

 This is the macro:
 macro dial-out(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;
 }


 I don't know if this is maybe related to this:
 https://issues.asterisk.org/jira/browse/ASTERISK-20441

 If it is this is a bug in the AEL compiler I think.


 --
 Johan Wilfer

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Re: [asterisk-users] How to log caller IP address in the CDR?

[Rafael dos Santos Saraiva]

Hi


Set(CDR(userfield)=${SIPPEER(${CALLERID(num),ip)})

If caller is SIP peer.


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-14 14:10 GMT-03:00 Rafael rrich...@gmail.com:


 can you please tell me exactly which file to edit please.



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[asterisk-users] CDR(dst) not set in AEL macro

[Rafael dos Santos Saraiva]

Hi

I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field  CDR(dst), showing only ~~s~~.

I tried various configurations, but without solutions.

This is the macro:
macro dial-out(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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[asterisk-users] CDR(dst) in AEL macro

[Rafael dos Santos Saraiva]

Hi

I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field  CDR(dst), showing only ~~s~~.

I tried various configurations, but without solutions.

This is the macro:
macro dial-out(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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Re: [asterisk-users] Sippeers realtime with minimum table

[Rafael dos Santos Saraiva]

Hi Joshua

I've tried to create a view in a database, but Asterisk requires updatable
fields in sippeers table (view), I cannot edit the structure of the tables.
I chose for create a php script to read database and create the sip.conf
file.

Thank's




Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-07-02 10:03 GMT-03:00 Joshua Colp jc...@digium.com:

 Rafael dos Santos Saraiva wrote:

 Hi there


 Kia ora,

  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.


 The field names have to be what chan_sip is expecting. If they don't
 match, then no go. One way to achieve this could be a view in your database.

 Cheers,

 --
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 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Sippeers realtime with minimum table

[Rafael dos Santos Saraiva]

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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Re: [asterisk-users] func_odbc

[Rafael dos Santos Saraiva]

The default method to include is #include, follow my func_odbc.conf
#include /opt/ipbx/asterisk/func_odbc.conf

[AGENDA]
dsn=asterisk
readsql=SELECT * FROM contacts ORDER BY code ASC
mode=multirow


And the included file /opt/ipbx/asterisk/func_odbc.conf:
[BLACKLIST]
dsn=asterisk
readsql=SELECT COUNT(*) AS count FROM blacklist WHERE
(calleridnum='${ARG1}' OR calleridnum = NULL) AND (dest='${ARG2}' OR dest =
NULL)



Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-04-03 20:32 GMT-03:00 Bryant Zimmerman brya...@zktech.com:

 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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[asterisk-users] Function REGEX

[Rafael dos Santos Saraiva]

Hi

I need help to use the function REGEX. My question is 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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Re: [asterisk-users] Function REGEX

[Rafael dos Santos Saraiva]

All working fine.
Thank you for your help.


Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230


2014-03-31 12:29 GMT-03:00 Eric Wieling ewiel...@nyigc.com:

 Here is an example from one of my production dialplans

 same =
 n,ExecIf(${REGEX(^1205|^1256|^1850|^1718|^1212|^1917|^1347|^1646|^1929
 ${CALLERID(num)})}]?Hangup)

 Assuming you meant 0-9 and not the literal X (which means nothing special
 in regular expressions):

 same = n,ExecIf(${REGEX(^[0-9]5123$ ${EXTEN})}]?Hangup)


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael dos Santos
 Saraiva
 Sent: Monday, March 31, 2014 11:23 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Function REGEX

 Hi

 I need help to use the function REGEX. My question is 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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[asterisk-users] Verbose only one context

[Rafael dos Santos Saraiva]

Hi

It's possible in Asterisk 1.8 enable verbose only in one context or
extension?

thanks

Att,
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http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
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[asterisk-users] Reverse Charging Indication MFCR2

[Rafael dos Santos Saraiva]

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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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

I noticed that the problem occurs when I use the variables ${VM_DUR} and
${VM_CALLERID}. Only the subject of the message, if the body is not the
problem. Using UTF or utf the same problem occurs.


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


2013/8/6 jg webaccou...@jgoettgens.de

 I checked the raw text of my voicemail messages today and I saw pretty
 much the same escape sequences for UTF-8 like you did, but I do not have
 any display problem. You could save the message locally and hand edit it
 (starting with uppercase UTF instead of lowercase utf).

 jg

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[asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

Hi

I have a problem w/ voicemail, the subject message is corruption when 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)*
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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

I tried with utf-8, iso8859-1 and us-ascii.

I used the Sendmail client, but now testing with mailcmd=cat 
/tmp/voicemail.txt

The version of Asterisk is 1.8.22.0.


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


2013/7/25 jg webaccou...@jgoettgens.de

 What is the value of charset in voicemail.conf?

 Have you tried a different Email client?

 jg

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Re: [asterisk-users] Voicemail variables on email subject

[Rafael dos Santos Saraiva]

When sending by SendMail the problem is the same in any email client.


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


2013/7/25 jg webaccou...@jgoettgens.de

 If I read your mailcmd correctly you are not really mailing but just
 dumping the data. Is the display correct when you use the default setting
 /usr/sbin/sendmail -t? You could send the mail to a local account and
 open it with mutt.

 jg

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[asterisk-users] Performance Asterisk large installation on Vmware/Xen

[Rafael dos Santos Saraiva]

Hi

I would like the opinion of you and if anyone has a similar scenario. I
have a project for installation of a Asterisk server in a client with about
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
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[asterisk-users] Extensions mask as variable?

[Rafael dos Santos Saraiva]

Hi,
How to define a extension mask as global variable in Ast 1.8? For example:
[globals]
MYVARIABLE = _15[7-9]X

I tried this way but it did not work.

Thanks
Att,
Rafael Saraiva
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[asterisk-users] Asterisk + Google Voice

[Rafael dos Santos Saraiva]

Hi
Is possible make calls from Asterisk with Google Voice?
The settings are done in jabber.conf and gtalk.conf? I was able to receive
calls from Gtalk on my Asterisk, but I would also like to generate calls to
the PSTN via Google.

Thank's
Rafael Saraiva
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Re: [asterisk-users] Asterisk + Google Voice

[Rafael dos Santos Saraiva]

Perfectly works!!

Thanks

Rafael Saraiva

2012/7/28 Matthew Jordan mjor...@digium.com


 - Original Message -

  From: Rafael dos Santos Saraiva rafaels...@gmail.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Saturday, July 28, 2012 9:18:56 AM
  Subject: [asterisk-users] Asterisk + Google Voice

  Hi
  Is possible make calls from Asterisk with Google Voice?
  The settings are done in jabber.conf and gtalk.conf? I was able to
  receive calls from Gtalk on my Asterisk, but I would also like to
  generate calls to the PSTN via Google.

 For Asterisk 11:

 https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google

 For older versions of Asterisk:

 https://wiki.asterisk.org/wiki/display/AST/Old+Calling+using+Google

 --
 Matthew Jordan
 Digium, Inc. | Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

Richard


I tried this, but it did not work. What can be the problem?
[PABX]
exten = _x.,1,Proceeding()
same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block)
same = n(allow),Dial(SIP/1584,30,tT))
same = n(block),Hangup()

Att,
Rafael Saraiva




2012/2/15 Richard Mudgett rmudg...@digium.com

   How to block collect calls on ISDN trunk?
 
  You need Asterisk v1.8 or later and check the value of
  CHANNEL(reversecharge) in your dialplan.
 
  https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL

  Can you give me an example of how to use this function?

 exten = 100,1,Proceeding()
 same = n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
 same = n(allow),Dial()
 same = n(block),Hangup()

 Please note that CHANNEL(reversecharge) is only valid on ISDN channels.

 Richard

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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

This is a variable received from the isdn channel.

Att,
Rafael Saraiva




2012/2/17 Danny Nicholas da...@debsinc.com

 Did you set CHANNEL(reversecharge) somewhere?

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 10:26 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

 ** **

 Richard

 ** **

 ** **

 I tried this, but it did not work. What can be the problem?
 

 [PABX]

 exten = _x.,1,Proceeding()

 same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block)

 same = n(allow),Dial(SIP/1584,30,tT))

 same = n(block),Hangup()

 ** **

 Att,
 Rafael Saraiva



 

 2012/2/15 Richard Mudgett rmudg...@digium.com

   How to block collect calls on ISDN trunk?
 
  You need Asterisk v1.8 or later and check the value of
  CHANNEL(reversecharge) in your dialplan.
 
  https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL

  Can you give me an example of how to use this function?

 exten = 100,1,Proceeding()
 same = n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
 same = n(allow),Dial()
 same = n(block),Hangup()

 Please note that CHANNEL(reversecharge) is only valid on ISDN channels.

 Richard

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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

The value is always -1. I must enable something in chan_dahdi to pass the
correct value?

++
[PABX]
exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
?entrada,${EXTEN},1:hangup,${EXTEN},1)
+++

rssr305*CLI -- Accepting call from '5132083300' to '1584' on
channel 0/18, span 1
-- Accepting call from '5132083300' to '1584' on channel 0/18, span 1
rssr305*CLI -- Executing [1584@PABX:1]
GotoIf(DAHDI/i1/5132083300-4, [-1 = -1] ?entrada,1584,1:hangup,1584,1)
in new stack
-- Goto (entrada,1584,1)
-- Executing [1584@PABX:1] *GotoIf(DAHDI/i1/5132083300-4, [-1 = -1]
?entrada,1584,1:hangup,1584,1*) in new stack
-- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
new stack
-- Goto (entrada,1584,1)
-- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
new stack
rssr305*CLI -- Executing [1584@entrada:2]
Dial(DAHDI/i1/5132083300-4, SIP/1584,30,tT) in new stack
-- Executing [1584@entrada:2] Dial(DAHDI/i1/5132083300-4,
SIP/1584,30,tT) in new stack
rssr305*CLI   == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
rssr305*CLI -- Called SIP/1584
-- Called SIP/1584
rssr305*CLI -- SIP/1584-001e is ringing
-- SIP/1584-001e is ringing
rssr305*CLI -- SIP/1584-001e answered DAHDI/i1/5132083300-4
-- SIP/1584-001e answered DAHDI/i1/5132083300-4
rssr305*CLI -- Span 1: Channel 0/18 got hangup request, cause 0
-- Span 1: Channel 0/18 got hangup request, cause 0
rssr305*CLI   == Spawn extension (entrada, 1584, 2) exited non-zero on
'DAHDI/i1/5132083300-4'
  == Spawn extension (entrada, 1584, 2) exited non-zero on
'DAHDI/i1/5132083300-4'
rssr305*CLI -- Hungup 'DAHDI/i1/5132083300-4'
-- Hungup 'DAHDI/i1/5132083300-4'
rssr305*CLI

Att,
Rafael Saraiva




2012/2/17 Danny Nicholas da...@debsinc.com

 I would put a Verbose statement after Proceeding to verify the value
 returned from ISDN channel, like this:

 **-  **Same = n,Verbose(RC value ${CHANNEL(reversecharge)})

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 11:07 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

 ** **

 This is a variable received from the isdn channel. 


 Att,
 Rafael Saraiva



 

 2012/2/17 Danny Nicholas da...@debsinc.com

 Did you set CHANNEL(reversecharge) somewhere?

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 10:26 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

  

 Richard

  

  

 I tried this, but it did not work. What can be the problem?
 

 [PABX]

 exten = _x.,1,Proceeding()

 same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block)

 same = n(allow),Dial(SIP/1584,30,tT))

 same = n(block),Hangup()

  

 Att,
 Rafael Saraiva


 

 2012/2/15 Richard Mudgett rmudg...@digium.com

   How to block collect calls on ISDN trunk?
 
  You need Asterisk v1.8 or later and check the value of
  CHANNEL(reversecharge) in your dialplan.
 
  https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL

  Can you give me an example of how to use this function?

 exten = 100,1,Proceeding()
 same = n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
 same = n(allow),Dial()
 same = n(block),Hangup()

 Please note that CHANNEL(reversecharge) is only valid on ISDN channels.

 Richard

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 ** **

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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

Which version do you recommend? Mine is 1.4.12.

Att,
Rafael Saraiva




2012/2/17 Danny Nicholas da...@debsinc.com

 From what I read, your libpri may be out of date.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 11:21 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

 ** **

 The value is always -1. I must enable something in chan_dahdi to pass the
 correct value?

 ** **

 ++

 [PABX]

 exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
 ?entrada,${EXTEN},1:hangup,${EXTEN},1)

 +++

 ** **

 rssr305*CLI -- Accepting call from '5132083300' to '1584' on
 channel 0/18, span 1

 -- Accepting call from '5132083300' to '1584' on channel 0/18, span 1*
 ***

 rssr305*CLI -- Executing [1584@PABX:1]
 GotoIf(DAHDI/i1/5132083300-4, [-1 = -1] ?entrada,1584,1:hangup,1584,1)
 in new stack

 -- Goto (entrada,1584,1)

 -- Executing [1584@PABX:1] *GotoIf(DAHDI/i1/5132083300-4, [-1 =
 -1] ?entrada,1584,1:hangup,1584,1*) in new stack

 -- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
 new stack

 -- Goto (entrada,1584,1)

 -- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
 new stack

 rssr305*CLI -- Executing [1584@entrada:2]
 Dial(DAHDI/i1/5132083300-4, SIP/1584,30,tT) in new stack

 -- Executing [1584@entrada:2] Dial(DAHDI/i1/5132083300-4,
 SIP/1584,30,tT) in new stack

 rssr305*CLI   == Using SIP RTP CoS mark 5

   == Using SIP RTP CoS mark 5

 rssr305*CLI -- Called SIP/1584

 -- Called SIP/1584

 rssr305*CLI -- SIP/1584-001e is ringing

 -- SIP/1584-001e is ringing

 rssr305*CLI -- SIP/1584-001e answered DAHDI/i1/5132083300-4**
 **

 -- SIP/1584-001e answered DAHDI/i1/5132083300-4

 rssr305*CLI -- Span 1: Channel 0/18 got hangup request, cause 0**
 **

 -- Span 1: Channel 0/18 got hangup request, cause 0

 rssr305*CLI   == Spawn extension (entrada, 1584, 2) exited non-zero
 on 'DAHDI/i1/5132083300-4'

   == Spawn extension (entrada, 1584, 2) exited non-zero on
 'DAHDI/i1/5132083300-4'

 rssr305*CLI -- Hungup 'DAHDI/i1/5132083300-4'

 -- Hungup 'DAHDI/i1/5132083300-4'

 rssr305*CLI 

 ** **

 Att,
 Rafael Saraiva



 

 2012/2/17 Danny Nicholas da...@debsinc.com

 I would put a Verbose statement after Proceeding to verify the value
 returned from ISDN channel, like this:

 -  Same = n,Verbose(RC value ${CHANNEL(reversecharge)})

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 11:07 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

  

 This is a variable received from the isdn channel. 


 Att,
 Rafael Saraiva


 

 2012/2/17 Danny Nicholas da...@debsinc.com

 Did you set CHANNEL(reversecharge) somewhere?

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael dos Santos
 Saraiva
 *Sent:* Friday, February 17, 2012 10:26 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Block Collect Calls on ISDN trunk

  

 Richard

  

  

 I tried this, but it did not work. What can be the problem?
 

 [PABX]

 exten = _x.,1,Proceeding()

 same = n,GotoIf($[${CHANNEL(reversecharge)} =-1]?allow:block)

 same = n(allow),Dial(SIP/1584,30,tT))

 same = n(block),Hangup()

  

 Att,
 Rafael Saraiva

 

 2012/2/15 Richard Mudgett rmudg...@digium.com

   How to block collect calls on ISDN trunk?
 
  You need Asterisk v1.8 or later and check the value of
  CHANNEL(reversecharge) in your dialplan.
 
  https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL

  Can you give me an example of how to use this function?

 exten = 100,1,Proceeding()
 same = n,GotoIf($[${CHANNEL(reversecharge)} = -1]?allow:block)
 same = n(allow),Dial()
 same = n(block),Hangup()

 Please note that CHANNEL(reversecharge) is only valid on ISDN channels.

 Richard

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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

Reversecharge not appear in debug.
I'm in Brazil, the signaling is different here?

Att,
Rafael Saraiva




2012/2/17 Richard Mudgett rmudg...@digium.com

  The value is always -1. I must enable something in chan_dahdi to pass
  the correct value?
 
 
  ++
 
  [PABX]
  exten=_X.,1,Gotoif([${CHANNEL(reversecharge)} = -1]
  ?entrada,${EXTEN},1:hangup,${EXTEN},1)
  +++
 
 
 
  rssr305*CLI -- Accepting call from '5132083300' to '1584' on
  channel 0/18, span 1
  -- Accepting call from '5132083300' to '1584' on channel 0/18, span 1
  rssr305*CLI -- Executing [1584@PABX:1]
  GotoIf(DAHDI/i1/5132083300-4, [-1 = -1]
  ?entrada,1584,1:hangup,1584,1) in new stack
  -- Goto (entrada,1584,1)
  -- Executing [1584@PABX:1] GotoIf(DAHDI/i1/5132083300-4, [-1 = -1]
  ?entrada,1584,1:hangup,1584,1 ) in new stack
  -- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
  new stack
  -- Goto (entrada,1584,1)
  -- Executing [1584@entrada:1] Answer(DAHDI/i1/5132083300-4, ) in
  new stack
  rssr305*CLI -- Executing [1584@entrada:2]
  Dial(DAHDI/i1/5132083300-4, SIP/1584,30,tT) in new stack
  -- Executing [1584@entrada:2] Dial(DAHDI/i1/5132083300-4,
  SIP/1584,30,tT) in new stack
  rssr305*CLI == Using SIP RTP CoS mark 5
  == Using SIP RTP CoS mark 5
  rssr305*CLI -- Called SIP/1584
  -- Called SIP/1584
  rssr305*CLI -- SIP/1584-001e is ringing
  -- SIP/1584-001e is ringing
  rssr305*CLI -- SIP/1584-001e answered DAHDI/i1/5132083300-4
  -- SIP/1584-001e answered DAHDI/i1/5132083300-4
  rssr305*CLI -- Span 1: Channel 0/18 got hangup request, cause 0
  -- Span 1: Channel 0/18 got hangup request, cause 0
  rssr305*CLI == Spawn extension (entrada, 1584, 2) exited
  non-zero on 'DAHDI/i1/5132083300-4'
  == Spawn extension (entrada, 1584, 2) exited non-zero on
  'DAHDI/i1/5132083300-4'
  rssr305*CLI -- Hungup 'DAHDI/i1/5132083300-4'
  -- Hungup 'DAHDI/i1/5132083300-4'
  rssr305*CLI


 The CHANNEL(reversecharge) value is set from the Reverse Charging
 Indication
 ie received in the incoming SETUP message.  Please capture the incoming
 SETUP
 from libpri for the collect call.
 pri set debug on span x

 Richard

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[asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

How to block collect calls on ISDN trunk?

Thank's

Att,
Rafael Saraiva
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Re: [asterisk-users] Block Collect Calls on ISDN trunk

[Rafael dos Santos Saraiva]

Richard
Can you give me an example of how to use this function?


Att,
Rafael Saraiva




2012/2/15 Richard Mudgett rmudg...@digium.com

  How to block collect calls on ISDN trunk?

 You need Asterisk v1.8 or later and check the value of
 CHANNEL(reversecharge) in your dialplan.

 https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL

 Richard

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[asterisk-users] calleridname presentation Asterisk = Siemens

[Rafael dos Santos Saraiva]

Hi

I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't
show the callerid name in the way Asterisk == Siemens. I realized that
Asterisk send calleridname in format namePresentationAllowedSimple to
Siemens e Siemens send calleridname in format
namePresentationAllowedExtended. I need change the format of the
calleridname in asterisk.

How to change?

Thank´s and sorry for my wrong english.
Att,
Rafael Saraiva
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Re: [asterisk-users] calleridname presentation Asterisk = Siemens

[Rafael dos Santos Saraiva]

Hi

I change for first way in Asterisk 1.8:

[teste]
include=rota00
exten=1504,1,Set(CALLERID(name-charset)=unknown)
exten=1504,2,Dial(DAHDI/g1/${EXTEN},60,tTwW)
exten=1504,3,Hangup()

But, in debug of the span show the simple form:

1   namePresentationAllowedSimple Context Specific [0 0x00] =
1 52 61 66 61 65 6C - Rafael

Att,
Rafael Saraiva



2011/7/1 Richard Mudgett rmudg...@digium.com

  I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i
  can't show the callerid name in the way Asterisk == Siemens. I
  realized that Asterisk send calleridname in format
  namePresentationAllowedSimple to Siemens e Siemens send calleridname
  in format namePresentationAllowedExtended. I need change the format
  of the calleridname in asterisk.
 
 
  How to change?

 There are two ways:

 1) With Asterisk v1.8 change the character set of the name to
 CALLERID(name-charset)=unknown.
 The default character set of iso8859-1 uses the simple form since that is
 the default character set of the extended form.

 2) Change libpri as follows in the function rose_enc_qsig_Name() to always
 send the extended form:
 --- rose_qsig_name.c(revision 2267)
 +++ rose_qsig_name.c(working copy)
 @@ -94,22 +94,12 @@
/* Do not encode anything */
break;
case 1: /* presentation_allowed */
 -   if (name-char_set == 1) {
 -   ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
 ASN1_CLASS_CONTEXT_SPECIFIC | 0,
 -   name-data, name-length));
 -   } else {
 -   ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos,
 end,
 -   ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
 -   }
 +   ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
 +   ASN1_CLASS_CONTEXT_SPECIFIC | 1, name));
break;
case 2: /* presentation_restricted */
 -   if (name-char_set == 1) {
 -   ASN1_CALL(pos, asn1_enc_string_bin(pos, end,
 ASN1_CLASS_CONTEXT_SPECIFIC | 2,
 -   name-data, name-length));
 -   } else {
 -   ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos,
 end,
 -   ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
 -   }
 +   ASN1_CALL(pos, rose_enc_qsig_NameSet(ctrl, pos, end,
 +   ASN1_CLASS_CONTEXT_SPECIFIC | 3, name));
break;
case 3: /* presentation_restricted_null */
ASN1_CALL(pos, asn1_enc_null(pos, end,
 ASN1_CLASS_CONTEXT_SPECIFIC | 7));

 Richard

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[asterisk-users] Conference feature

[Rafael dos Santos Saraiva]

Hi

How to create the conference feature in Asterisk?

Thank's

Att,
Rafael Saraiva
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Re: [asterisk-users] Conference feature

[Rafael dos Santos Saraiva]

I am referring to 3-way conference

Att,
Rafael Saraiva



2011/6/26 Flavio Miranda flaviormira...@hotmail.com


 Very simple..

 Just edit the meetme.conf in /etc/asterisk like this :
 [rooms]

 conf = 888

 And then, in /etc/asterisk/ extensions.conf , put something like that:

 [conference]

 exten = 888,1,Set(CHANNEL(language)=pt_BR)if you have pt_BR audio
 exten = 888,n,MeetMe(888,pdM)
 exten = 888,n,Playback(vm-goodbye)
 exten = 888,n,Hangup

 When an user call 888 he will be in a conference  room.

 I hope it  help!


 Att,

 Flavio Roberto Miranda
 MSN:flaviormira...@hotmail.com
 Skype: flaviormiranda

 --
 Date: Sun, 26 Jun 2011 22:25:00 -0300
 From: rafaels...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Conference feature

 Hi

 How to create the conference feature in Asterisk?

 Thank's

 Att,
 Rafael Saraiva


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[asterisk-users] calleridname presentation Asterisk == Siemens

[Rafael dos Santos Saraiva]

Hi

I interconnect the Asterisk and the Siemens PBX with Pri QSIG. But i can't
show the callerid name in the way Asterisk == Siemens. I realized that
Asterisk send calleridname in format namePresentationAllowedSimple to
Siemens e Siemens send calleridname in format
namePresentationAllowedExtended. I need change the format of the
calleridname in asterisk.

How to change?

Thank´s and sorry for my wrong english.
Att,
Rafael Saraiva
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Re: [asterisk-users] DAHDI span timeing source

[Rafael dos Santos Saraiva]

Hi

The timing source is the clock of the system. When a equipment is 0, the
other should be 1. The correct is: 0=slave, 1=master. The default for
private systems is slave.

Att,
Rafael Saraiva

2011/5/27 satish patel satish...@hotmail.com

  Hi There,

 We have very old asterisk 1.2 running in production and it has following
 setting in /etc/zaptel.conf.  I have read on web about span and they told
 span= span num ,timing source,line build out
 (LBO),framing,coding[,yellow]

 Just wondering why it has timing source 0 ?  0=master, 1=slave  right ? Do
 you think i should change it to 1 ?

 #Sangoma A102 port 1 [slot:2 bus:7 span:1] wanpipe1
 span=1,0,0,esf,b8zs
 bchan=1-23
 dchan=24

 #Sangoma A102 port 2 [slot:2 bus:7 span:2] wanpipe2
 span=2,0,0,esf,b8zs
 bchan=25-47
 dchan=48


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Re: [asterisk-users] DAHDI span timeing source

[Rafael dos Santos Saraiva]

Really, You're right. This option define the priority of the interface as
regenerator of clock:
priority 0 = its own clock
priority 1 = the clock of the telco



2011/5/27 Satish Patel satish...@hotmail.com

 I guess you are wrong here correct one is 0=master 1=slave

 If you connect to PSTN the you should user span=1,1,0

 Check out   http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html
 http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html

 --
 Sent from my iPhone

 On May 27, 2011, at 4:27 PM, Rafael dos Santos Saraiva 
 rafaels...@gmail.com wrote:

 Hi

 The timing source is the clock of the system. When a equipment is 0, the
 other should be 1. The correct is: 0=slave, 1=master. The default for
 private systems is slave.

 Att,
 Rafael Saraiva

 2011/5/27 satish patel  satish...@hotmail.comsatish...@hotmail.com

  Hi There,

 We have very old asterisk 1.2 running in production and it has following
 setting in /etc/zaptel.conf.  I have read on web about span and they told
 span= span num ,timing source,line build out
 (LBO),framing,coding[,yellow]

 Just wondering why it has timing source 0 ?  0=master, 1=slave  right ? Do
 you think i should change it to 1 ?

 #Sangoma A102 port 1 [slot:2 bus:7 span:1] wanpipe1
 span=1,0,0,esf,b8zs
 bchan=1-23
 dchan=24

 #Sangoma A102 port 2 [slot:2 bus:7 span:2] wanpipe2
 span=2,0,0,esf,b8zs
 bchan=25-47
 dchan=48


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Re: [asterisk-users] Pridialplan/ prilocaldialplan

[Rafael dos Santos Saraiva]

Hi
I made a mistake. I was putting the lines pridialplan and
prilocaldialplan after
the line channel in chan_dahdi.conf. The Asterisk does not read the
lines after channel.

Thank's

Att,
Rafael Saraiva


2011/5/23 Rafael dos Santos Saraiva rafaels...@gmail.com

 did not work!! Bug in Asterisk?? :(

 Rafael

 2011/5/20 Захаров Антон ins...@mail.ru

  Yeap, I couldn't set Private TON too. Try to set all _prefix variables in
 chan_dahdi.conf and use dynamic prilocaldialplan.

 On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote:

 Hi

  I change the chan_dahdi.conf and restart dahdi:
 prilocaldialplan=private
 pridialplan=private

  But, in debug i see the following informations:
  1  Calling Number (len= 8) [ Ext: 0  TON: *National Number (2) * NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 1Presentation: Presentation permitted, user
 number not screened (0)  '1570' ]
 1  [70 0a 80 30 38 31 37 34 37 39 35 36]
 1  Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
 Unknown Number Plan (0)  '81747956' ]

  I set Private TON, but display National TON.

  Thank's

  Att,
 Rafael Saraiva
 2011/5/19 Захаров Антон ins...@mail.ru

  Hello.

 To apply this settings you should restart dahdi (dahdi restart in CLI).
 About influence you could read here:
 http://markmail.org/message/rpd2aewiu2soostz

 On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote:

 Hi


  I'm beginner in list. I have doubts about the options pridialplan and
 prilocaldiaplan 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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Re: [asterisk-users] Pridialplan/ prilocaldialplan

[Rafael dos Santos Saraiva]

did not work!! Bug in Asterisk?? :(

Rafael

2011/5/20 Захаров Антон ins...@mail.ru

  Yeap, I couldn't set Private TON too. Try to set all _prefix variables in
 chan_dahdi.conf and use dynamic prilocaldialplan.

 On 19.05.2011 21:30, Rafael dos Santos Saraiva wrote:

 Hi

  I change the chan_dahdi.conf and restart dahdi:
 prilocaldialplan=private
 pridialplan=private

  But, in debug i see the following informations:
  1  Calling Number (len= 8) [ Ext: 0  TON: *National Number (2) * NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 1Presentation: Presentation permitted, user
 number not screened (0)  '1570' ]
 1  [70 0a 80 30 38 31 37 34 37 39 35 36]
 1  Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
 Unknown Number Plan (0)  '81747956' ]

  I set Private TON, but display National TON.

  Thank's

  Att,
 Rafael Saraiva
 2011/5/19 Захаров Антон ins...@mail.ru

  Hello.

 To apply this settings you should restart dahdi (dahdi restart in CLI).
 About influence you could read here:
 http://markmail.org/message/rpd2aewiu2soostz

 On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote:

 Hi


  I'm beginner in list. I have doubts about the options pridialplan and
 prilocaldiaplan 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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 Rafael Saraiva


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Re: [asterisk-users] how to user SIP realtime option

[Rafael dos Santos Saraiva]

Hi
Trying exclude rtcachefriends from your sip.conf and include the field
rtchachefriends in table sip_buddies. And exclude the field qualify from
sip_buddies. Set YES in field rtcachefriends.

Att,
Rafael Saraiva

2011/5/21 virendra bhati virbh...@gmail.com

 Hi List,

 After read the link
 http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . I changes all
 information in below conf files

 *res_mysql.conf*

 [mpathsala]
 dbhost = localhost
 dbname = mpathsala
 dbuser = root
 dbpass =
 dbport = 3306

 *extconfig.conf*

 sipusers = mysql,mpathsala,sip_buddies
 sippeers = mysql,mpathsala,sip_buddies

 *mysql*

 make same table sip_buddies in mpathsala database.
 insert 1 row into database

 [image: Full 
 Texts]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiessql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560dontlimitchars=1token=0ba5913eba31b6657e85f0a1ffec66de
  id
 name host nat type accountcode amaflags call-limit callgroup callerid 
 cancallforward
 canreinvite context defaultip dtmfmode fromuser fromdomain insecure language
 mailbox md5secret deny permit mask musiconhold pickupgroup qualify regexten
 restrictcid rtptimeout rtpholdtimeout secret setvar disallow allow fullcontact
 ipaddr port regserver regseconds lastms username defaultuser subscribecontext
 useragent [image: 
 Edit]http://192.168.193.69/phpmyadmin/tbl_change.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66deprimary_key=+%60sip_buddies%60.%60id%60+%3D+1sql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php
   [image:
 Delete]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66desql_query=DELETE+FROM+%60sip_buddies%60+WHERE+%60sip_buddies%60.%60id%60+%3D+1+LIMIT+1zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dmpathsala%26table%3Dsip_buddies%26token%3D0ba5913eba31b6657e85f0a1ffec66de%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dsql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560
 1 300 dynamic no friend *NULL* *NULL* *NULL* *NULL* 100 yes yes *NULL* *
 NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL
 * *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* all
 g729;ilbc;gsm;ulaw;alaw 0 *NULL* 0 0 *NULL* *NULL*

 *sip.conf *

 [general]

 rtcachefriends=yes
 rtsavesysname=yes
 rtupdate=yes
 rtautoclear=yes

 After all when I check on CLI then I will get

 *cent70*CLI sip show peers*
 Name/username  HostDyn Nat ACL Port Status
 Realtime
 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
 offline]

 *cent70*CLI sip show users*
 Username   Secret   Accountcode
 Def.Context  ACL  NAT

 Why SIP/300 is not display here ??

 Please help me I want to learn asterisk real-time concept to make my server
 real-time.
 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Asterisk Engineer


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Re: [asterisk-users] Pridialplan/ prilocaldialplan

[Rafael dos Santos Saraiva]

Hi

I change the chan_dahdi.conf and restart dahdi:
prilocaldialplan=private
pridialplan=private

But, in debug i see the following informations:
1  Calling Number (len= 8) [ Ext: 0  TON: *National Number (2) * NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
1Presentation: Presentation permitted, user
number not screened (0)  '1570' ]
1  [70 0a 80 30 38 31 37 34 37 39 35 36]
1  Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0)  '81747956' ]

I set Private TON, but display National TON.

Thank's

Att,
Rafael Saraiva
2011/5/19 Захаров Антон ins...@mail.ru

  Hello.

 To apply this settings you should restart dahdi (dahdi restart in CLI).
 About influence you could read here:
 http://markmail.org/message/rpd2aewiu2soostz

 On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote:

 Hi


  I'm beginner in list. I have doubts about the options pridialplan and
 prilocaldiaplan 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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[asterisk-users] Pridialplan/ prilocaldialplan

[Rafael dos Santos Saraiva]

Hi


I'm beginner in list. I have doubts about the options pridialplan and
prilocaldiaplan 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
-- 
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Rafael Saraiva
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