similar to: Agent Login "Play a file"

According to the instructions given at https://www.asterisksounds.org/de I converted and installed German prompts successfully and for numbers, I can successfully listen to a German female voice counting or telling the date/time. But unlikily, somehow the voicemail prompt is still English, although my general language settings are "de". I use pjsip.conf, not sip.conf. In

Hello, i have to write a script to visualize the output of tc -s class show dev eth0 with rrdtool. Or does anybody know of such a script , which is available for download ? I assume the bps in "rate 5728bps 34pps" is Byte per Second. Is that right ? How can i set all the counters back to zero ? I did not found anything in the manpage of tc. regards Joerg -- Jörg Hartmann Tel:

Hi All, I'm going to be upgrading our 1.2 Asterisk system. At the moment we use the Enicomms SLN files. Are there major differences in the 1.4 default voicefile packs, or will I be able to re-use Enicomms?? In the Make menuselect, I noticed theres no .SLN voicefile selection for the basic audiofiles - has SLN been depreciated? Thanks Adrian

Scenario: Our Asterisk 13 PBX (on network 192.168.254.0/24, bound to 192.168.254.1:5060) is behind a NAT, acting as a client to our ITSPs SIP server. But also, this Asterisk is server for various VoIP telephones. Acoording to Asterisk's wiki, the transport section of pjsip.conf is configured as follows: ; Transport via UDP [transport-nat-udp] type= transport

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello, we're running a small Asterisk appliance on a PCengine APU2C4. Base operating system is FreeBSD 12-STABLE, most recent incarnation as of today. Since update of port net/asterisk16 to the latest bug fix revision 16.6.1, we face a severe "slowdown" of everything that the Asterisk core performs, i.e. outgoing calls are delayed ~

Dear Sirs. After Upgrading and installing FreeBSD 5.1-R-p2 from FreeBSD 4.8 (completely fresh installation!) I can not do any manpages and can not do a make world anymore. Man pages would work when setting up an exported environment variable export GROFF_FONT_PATH=/usr/share/groff_font export GROFF_TMAC_PATH=/usr/share/tmac and appropriate in csh. I searched the mailing list and found a lot

Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but <10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems

+Snesha Foss <sneshaf@microsoft.com> who is taking over this work. It definitely looks like adding the query string to every path is just wrong. I'm not sure I understand why we'd want to parse, deconstruct key value pairs and then reconstruct the query string from these values and append them to the path selectively. This seems like added complexity for a benefit I don't

Dear Sirs. It seems to me a never ending story. We run a box with a TYAN Thunder 2500 Dual SMP mainboard, 2GB ECC Tyan certified memory, AMI Enterprise 1600 RAID adapter and additional Intel 1000/Pro server type (64 bit) GBit LAN NIC. With FreeBSD 4.8 this was stable, but to achive this state was really hard! It is a story similar to that what happend when we changed towards FreeBSD

I'm quite new to Asterisk and using Asterisk 13 on FreeBSD current. Asterisk is behind a NAT router, the physical setup is very much a trivial one. The Asterisk PBX is supposed to act as the telephone gateway for several VoIP/SIP phones. I'm using throughout pjsip as configuration, I have no experience with chan_sip since I started recently using Asterisk for several SoHo and lab's

Is for these UFS bugs in FreeBSD since 6.1 a fix uderway? See: http://projects.info-pull.com/mokb/ MOKB-08-11-2006,CVE-2006-5824, MOKB-03-11-2006,CVE-2006-5679 Regards, Oliver

Is for these UFS bugs in FreeBSD since 6.1 a fix uderway? See: http://projects.info-pull.com/mokb/ MOKB-08-11-2006,CVE-2006-5824, MOKB-03-11-2006,CVE-2006-5679 Regards, Oliver

Hi, I have a problem where I can join an xpsp2 machine to a domain but, no matter what %COMPUTERNAME% i use, it says "system error: a duplicate name exists on the network" after the reboot when upon successfully joining. If I try to log in as a valid user, i get the "the system could not log you on because domain 'DOMAIN' is not available". I'd just like to

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

Anyone know if it is possible to control how aggressively the "Aggressive" mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie

Hi all, I am not 100% sure if this is a -dev or a -user topic, but I am leaning towards the former. Feel free to cuss at me and tell me to ask -user, instead. I used to run a patchset that allowed full logs of everything taking place via OpenSSH. This also allowed me to replay any session, live or after the fact. I am fully aware of the security implications of logging everything, especially

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf

Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to