similar to: DBdeltree: Error deleting key from database

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 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(

Is it possible to delete the keys belonging to a family in AstDB at Asterisk startup? I would like to repopulate it from another source each time Asterisk is restarted. I know there is a DBdeltree(<family>) function. Is there a context that only runs once (automatically) at Asterisk startup (so that I can call this function)? Also is AstDB lookup faster than a func_odbc lookup? Is there a

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 guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

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=-

hi just donwloaded the 1.6.1 branch and made configure & install. so far so good. after staerting asterisk with: asterisk -vvvvcr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls == Manager registered action Park == Manager registered action Bridge == Manager registered

I think I have found the reason for this: The setup runs just fine until I set the xen dom0 to only use one of the four CPUs in my machine (actually 2 HT CPUs). So with (dom0-cpus 0) in /etc/xen/xend-config.sxp this works. The while-loop actually ran fine for 2 days straight. With (dom0-cpus 1) it crashes as described within a few minutes. I will cc this to the xen-list. Full thread here:

I think I have found the reason for this: The setup runs just fine until I set the xen dom0 to only use one of the four CPUs in my machine (actually 2 HT CPUs). So with (dom0-cpus 0) in /etc/xen/xend-config.sxp this works. The while-loop actually ran fine for 2 days straight. With (dom0-cpus 1) it crashes as described within a few minutes. I will cc this to the xen-list. Full thread here:

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 =>

I have installed the Perl library from http://asterisk.gnuinter.net/asterisk-perl and am wondering how I reference agi-test.agi from extensions.conf I have added exten => s,1,AGI,agi-test.agi but that doesn't seem to do it. Is there a certain directory .agi files should be, is that the problem? TYIA Dan

Config has ssl_verify_client_cert = no What options might have the client auth turned on? TYIA On 5/24/20 6:40 PM, Felipe Gasper wrote: > From what I can tell, ?SSL alert number 42? means that you?ve configured Dovecot to require client authentication. > > Otherwise, your Let?s Encrypt certificate (with its authority chain) should suffice. > > -FG > >> On May 24,

Hi, I'm attempting to join a Windows Server 2022 to an existing domain Im running into issues as I am trying to migrate away from Samba DCs (:sad:) I've been able to successfully join a Windows Server 2022 to a fresh domain without much trouble after following tranquil.it<https://samba.tranquil.it/doc/en/samba_advanced_methods/samba_add_windows_active_directory.html>'s guide as

Hello All, Eric Langley here with my first post to this list. I am looking to determine if R is suitable for a development project I am working on and if so possibly finding someone proficient in R that would be interested in doing the coding. I would like to preface my inquiry that while I am not a programmer I can communicate in a dialog my objectives. An array of rank ordered data looks like

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

I've been asked to create a graphical "call-queue" manager. That is, use the existing call queues application but allow a way to view what's coming and attach information to it. As far as the "attaching information" that's in the realm of my application, but I'm trying to figure out if the internals of queues are exposed through any interface. Any help there?

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:

I am running Poptop 1.1.4 VPN server, Samba 3.0.6, RH 9 (2.4.20-8 patched for mppe-mppc). When the end user connects via VPN, they can create a share to the samba shared folder that was created. When they clost the explorer window, then reopen the shared drive from the "My Computer" view, the share can not be re-opened with the error that the mapped drive is alreay in use. I have

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

I'm looking for some help from any Asterisk "heavy" who might be doing something similar to what I'm trying to do... Background: I work for a research lab, testing telephony products and tools. Historically, we used Ameritec Crescendos and Fortissimos to act as load generators and call "sinks" when testing equipment. However, the equipment we are testing gets