Archives : October-2014
while moving from asterisk 12.3 to asterisk 12.6, I see the MWI support for voicemail has stopped working. If I check sip show peer 104-DEVEL on asterisk 12.3, I can clearly see the Mailbox option set, while on asterisk 12.6 it appears empty.Is th..
The requested log is attached.In the last lines of the file there are the records of the call transfer:[Oct 17 17:52:14] VERBOSE[4314] cdr_adaptive_odbc.c:> [INSERTINTO cdr(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,uniqueid)VAL..
begin 644 cdrlog.txt M6T]C=`Q-R`Q-SHU,3HT-5T@5D520D]315LT,S$T72!C9(N8SH@(#!X,C8YM-V-C.`M($-R96%T96..
I wrote:Queue(ENGLISH,rh,,,20,,,log-answer)but no variables are available in the subroutine context. I need to get ${UNIQUEID} in there.In a Dial command, I can write U(log-answer^${UNIQUEID}) but not in the gosub field of the Queue command.Is it possi..
Hello I have a simple 1 channel goip gateway (http://www.voip-info.org/wiki/view/GoIP). The incoming and outgoing calls work with Asterisk except the caller ID for the outgoing calls. I think I have exhausted all possible options regarding settin..
Everyone!We wanted to let everyone coming to Astricon know that we will be holding an OpenSIPS Summit on Tuesday Oct 21st, 2014 at the Suncoast Casino & Spa.Suncoast is about 10 minutes away from Red Rock and we will be provide shuttle service to ..
,Im searching someone who already installed allo.com gsm card with Asterisk NOW.I installed the hardware in my new server, but when runing dahdi_genconf I always have the no span message. I tried to install the driver according to allo.com doc, but..
there, I have installed Asterisk version 12.6 (on Debian wheezy) and I note that, only when I make a transfer of call (attended or unattended), the fields dst and dcontex in the CDR are empty. This happen both in MySQL record and in CVS.Someone can conf..
I have a client that wants a phone system that will play sounds from asleep machine. I tried using all different formats (GSM, WAV, WV49,MP3 etc.). Over SIP it was OK however with the PSTN it broke up fromtime to time. I assume this has to do with ..
all,According to the documentation (http://www.voip-info.org/wiki/view/Asterisk+h+extension):Be aware: Macros require their own h extension as they do not make use of the calling contexts h extension!Does this apply to subroutines too? I am unable..