similar to: Asterisk 13 stable?

Sorry, I forget to tell I tried, but not works. *Context:* context ivr_temp2 { s => { Proceeding(); str_time_01 = '06:00-12:00|*|*|*'; // Manh? ifTime (${str_time_01}) { Playback(ura/bom_dia); } } } The error is showed on "ael reload". *Console errors:* rs0000sr304*CLI> ael reload Command 'ael reload' failed.

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>

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)

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 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

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)

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> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140326/4ed97cc9/attachment.html>

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> -------------- next part -------------- An HTML attachment was scrubbed... URL:

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; }

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?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 at sv03 asterisk]# ls -lahZ /var/lib/asterisk/astdb.sqlite3

Prezado Fernando, Muito obrigado por sua complementa??o na resposta! Surgiram algumas d?vidas agora: A ?nica forma de retornar os dados num header field, como o Rafael dos Santos Saraiva sugeriu envolve criar outro channel? Ou seja, o que eu preciso ? que a mesma execu??o do dia plan obtenha um valor recebido do Sip Client, execute uma query num banco de dados e em seguida inclua a resposta

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

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*

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

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 -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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> -------------- next part -------------- An HTML attachment was scrubbed... URL:

At any rate: Error(SUBJECT/IV) specifies two random effects: SUBJECT and SUBJECT:IV. This is most easily understood if you conceptually arrange your data in a SUBJECT x IV table: One effect is a set of random errors added to each row, the other is a set of effects added to each cell. If you have more than one observation within each cell, then you need a third set of errors to account for

Bert, thanks for the reply but I feel that my question is less about statistics and more about R interface. Specifically, because the output of R seems different than other programs (systat, for example, gives a between and a within table instead of a three level one). I am familiar with the connection between mixed models and repeated measures,and how mixed models are essentially replacing the