Archives : July-2015
I would like to figure out using confbridge how to play a file after the conf is built.not really a per user thing – just conf is up and ready and need to play a file to all in the conference.(I am creating my conf on the fly and bringing in other devi..
I am currently running Asterisk 13.1.0-1I have a chan_sip configuration that works fine with a 3rd party.Third party does not use authentication or registration, its ip based authentication…When I try switching to PJSIP.conf, I seeing 488 respon..
guys,there is something im not sure and would like to ask please :lets say i have B calling and talking to A.(A B)Then B would like to attended transfer A to C. In the behind extensions.conf when B is calling C (first part of the attended transfer) ..
Hi.I my dialplan I have :same = n,Dial(PJSIP/6001,10)same = n,Dial(PJSIP/6002,30)same = n,Hangup()The extension 6002 will not be inviteduntil the called party 6001 hangs up or until 10 seconds if nobody answers the call in 6001.How to call 6001 and immediat..
Im running MariaDB with Asterisk 1.8.11.0 over ODBC for CEL and CDRlogging.CDRs work fine, times logged in the DB are correct consistently.However, Ive noticed that for CELs eventtimes start lagging severely.E. g. Id start Asterisk at 12:15 and entr..
Hi.The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesnt explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too.So, can I ..
Hi.I have read in some web sites that ASTERISK can support messages out of calls. What does it exactly means?1 – Can a dialplan script accept and handle a message from a callee party, even before the call be connected?2 – Can a ringing callee send ..
again,Im trying to send two E-Mails when a message comes in the voicemail,the first WITH the attachment, and the second WITHOUT. But I dont get it working…I wrote00493511111111 => SECRET,John Doe,first@emailde,second@email.de,attach=nobut both E-M..
!I need a change on my Voicemail configuration, but I cant experimentnow, since the system is in use…I have some voicemail. The system has to send an E-Mail (with the WAVattached) to an address AND another E-Mail, without the WAV to anotheraddres..
Dear ASTERISK-users,What Dial Plan function can access the contents of the SDP ?If there is no Dial Plan Function for that, is there some another way to access contents of the SDP? Maybe via ARI ou AGI?If there is, how to access the SDP that comes w..