similar to: CDR(dst) not set in AEL macro

Displaying 20 results from an estimated 300 matches similar to: "CDR(dst) not set in AEL macro"

2014 Jul 10
0
CDR(dst) in AEL macro
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; }
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
2015 May 12
1
AEL keyword IfTime with variable on time range
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 (${str_time_01}) { Playback(ura/bom_dia); } } } The error is showed on "ael reload". *Console errors:* rs0000sr304*CLI> ael reload Command 'ael reload' failed.
2015 May 12
2
AEL keyword IfTime with variable on time range
Hi It's possible using a variable in the iftime keyword argument? E.g: context text { s => { timerange = '06:00-12:00|*|*|*'; ifTime(${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>
2018 Nov 30
2
Asterisk non-root - selinux - astdb
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
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> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140326/4ed97cc9/attachment.html>
2014 Mar 31
1
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*
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> -------------- next part -------------- An HTML attachment was scrubbed... URL:
2013 Aug 05
3
Voicemail variables on email subject
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)
2013 May 18
5
Performance Asterisk large installation on Vmware/Xen
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
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> -------------- next part -------------- An HTML attachment was scrubbed... URL:
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 -------------- next part -------------- An HTML attachment was scrubbed... URL:
2011 May 19
1
Pridialplan/ prilocaldialplan
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
2015 Jun 12
0
RES: Banco de dados interno no Asterisk e variáveis em SIP HEADERS
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
2015 Aug 12
2
Busy level in Asterisk 11
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)
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> -------------- next part -------------- An HTML attachment was scrubbed... URL:
2017 Dec 29
0
Why aov() with Error() gives three strata?
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
2017 Dec 28
0
Why aov() with Error() gives three strata?
Jorge: FYI, *generally speaking,* queries that are mostly statistical in nature, such as yours, are off topic here -- this list is about R programming help, not statistical help. Having said that, you still may get a useful response here -- the r-help/statistics intersection *is* nonempty. However, if not, 2.5 suggestions: 1. Try posting to r-sig-mixed-models instead. Repeated measures are a
2017 Dec 28
2
Why aov() with Error() gives three strata?
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
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