similar to: Set qualify = yes on trunk can't do outgoing call

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

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>

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 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, 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 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 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 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 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 -------------- next part -------------- An HTML attachment was

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 -------------- next part -------------- An HTML

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

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:

hello all, just to say that this days was the 6th international free soft forum in Porto Alegre Brasil, and they where streaming all the talks with ogg/theora format ... using icecast server and pd/pidip encoder. http://tv.softwarelivre.org/bin/view/TV/ thanks theora :)

Have someone experienced any strange noises using the ilbc codec after upgrading to asterisk 1.0.3? I had to change the codec do gsm to fix this problem. The noise is very loud, like saturation of the echo ro something, seems like the echo cancelation is amplifying itself. I'be been using ilbs since asterisl 0.70 and have never had any problem like this. Thanks. --

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 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; }

Is there any package similar to the XML package that is able to "extract" relevant information from HTML files. Namely, I'm interested in obtained data that is represented as a HTML table, into some R-type structure. Thank you. -- Luis Torgo FEP/LIACC, University of Porto Phone : (+351) 22 607 88 30 Machine Learning Group Fax : (+351) 22 600 36 54 R. Campo

Hi There, I am also trying to do the same for my network. I have two links from different ISPs and I want to configure a failover and load balancing Linux router. I am facing same problem here, that how to detect link failure and let Linux box switch the gateway. I know it works when the first gateway is physically down and not reachable. But what to do if my link is up but there is problem

Hi THere, sorry if this is a stupid question or does not belong to this forum. I''ve set my DEFROUTE=no in my ifcfg-ppp0 and when I bring the ppp0 up, it deletes my old default load balance routes which I do not want, as I just want the interface to be up, but not touching my default routes any advice Thanks SEW