search for: satish4asterisk

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi, I have small confusion in my configuration which is I had some DID's like 044578900-04457999. I was configured dial plan below mention. exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten =>

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have been working on a project where I need to record a call in Asterisk and then split the recording into multiple audio files based on a presence of particular sound (i.e. beep) in a recording. I know this is out of scope for Asterisk but I wanted to benefit from someone else's experience if it has been done earlier. I have googled a bit and seems that Audio fingerprint(

While playing with DB function in Dialplan, I have added some garbage in AstDB. These are some family names with space in them. See this, demo*CLI> database show /18-05-2011 00:00:0052011175221575/TESTDATE : 2011-05-14 21:33:46 /18-05-2011 00:00:0052011175221575/TEST1 : 410 /18-05-2011 00:00:0052011175221575/TEST2 : 155 /18-05-2011 00:00:0052011182614252/TEST3 :

Hi All, I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP. On incoming calls from Avaya asterisk complains of 'unsupported crypto parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable here' Doesn't Asterisk support UNENCRYPTED_SRTCP as crypto parameters in sdp? FYI SDP looks like this. v=0 o=- 1429194215 1 IN IP4 XX.XX.XX.XX s=-

Is there any option in Asterisk distribution of this? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120531/8fac6a22/attachment.htm>

I was looking for MySQL table structures for ARA (Asterisk 1.8.X). I found one for SIP friends on, https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime,+MySQL+table+structure But it seems that it is not as per the Asterisk 1.8 SIP options. i.e. it contains 'call-limit' which is deprecated in 1.8 and not the 'callcounter' as one of the fields. Pardon my ignorance, but are

Hi Everybody, Don't know why this DBdeltree error appears on Asterisk CLI.Good part is, it does remove family entry from AstDB. Sample Dialplan.... exten => 1212,1,Noop() same => n,Set(TEST=1234) same => n,Set(DB(${TEST}/TESTSTART)=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)}) same => n,DBdeltree(${TEST}) same => n,Hangup() Asterisk CLI output.... [Jun 1 14:30:39] == Using

Hi, I have 2 Asterisk systems and a unique scenario where I need to play a particular tone on Asterisk1 and identify the same tone on Asterisk2. Following is my call flow, Asterisk1(Plays audiofile1,Wait for 2 Sec,Plays a tone,Plays audiofile2) -> PSTN -> 3rd Party CONFERENCE SYSTEM <- PSTN <- Asterisk2(Record audiofile1,Wait for a tone,Record audiofile2). A few points to keep in