similar to: Sippeers realtime with minimum table

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 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 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 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 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 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 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)

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 have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas

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

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, all after my installation of asterisk and adds-on . when start astrisk, error accours as follow: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available what's wrong with me ? Thanks. -- Best regards, Sucan

Hi Andrew, OK, (the simple fact that those machines are not connected to internet makes that i have to go to those machines and copy them on a usb-stick, so it causes some delay each time...) -------- Forwarded Message -------- Sorry - I meant extconfig.conf - not cdr_mysql.conf (my mistake). I use (and done for a long time) mySQL for realtime storage - and it's never let me down (touch

Still a couple of questions...... I did configure extconfig.conf ... ;iaxusers => odbc,asterisk ;iaxpeers => odbc,asterisk ;sipusers => odbc,asterisk sipusers => mysql,asterisk,sip_devices sippeers => mysql,asterisk,sip_devices ;sippeers => odbc,asterisk ;sipregs => odbc,asterisk ;voicemail => odbc,asterisk ;extensions => odbc,asterisk ;meetme => mysql,general

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*

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

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, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

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:

I'm running sippeers and sipusers in my extconfig, and everything runs perfectly when a client is registered (ex. registers to port 1000), but when it re-registers the client is set to port 5060. This behavior does not take place if I use the static files. Both in my sip_buddies table for db, and sip.conf for static I have host=dynamic and nat=yes. :M