similar to: Busy level in Asterisk 11

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 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:

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

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

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 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; }

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 am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

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:

I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for testing. In addition I register a zoiper SIP soft phone. For the Grandstream I have busylevel=1 in sip.conf. If I place a call from the GXP280 to zoiper and then put that call on hold from the zoiper side and then call GXP280's extension, asterisk indicates the phone is ringing. As the GXP280 is a single line phone it

I sent this email a few days ago but did not see any responses to it: > I am running asterisk 1.6.0.1. I have a Grandstream GXP280 phone I use for > testing. In addition I register a zoiper SIP soft phone. > > For the Grandstream I have busylevel=1 in sip.conf. > > If I place a call from the GXP280 to zoiper and then put that call on hold > from the zoiper side and then

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a ?crit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - --

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

Hi It seems, I just miss something. I defined treshold <- function(pred) { if (pred < 0.5) pred <- 0 else pred <- 1 return(pred) } and want to use apply it on a vector sapply(mylist[,,3],threshold) but I get: Error in match.fun(FUN) : Object "threshold" not found thanks for help cheers chris -- Christoph Lehmann <christoph.lehmann at gmx.ch>

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE-----