similar to: Converting dahdi_monitor unit to dbm0

I am studying about echo cancellation in asterisk and I want to use the numeric information from dahdi_monitor verbose for my research. Unfortunately, I couldn't find anything about the unit of measurement used in this tool. Which unit is used to measure the signal level? -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice

Hello, Can you please tell me what the overload point is for OPUS in dBm0 or dBV? As an example, the G.711 A law codec has an overload point of +3.14 dBm0 across a 600 ohm circuit. In ITU-T G.100.1 section 5.8 you can read more about the relationship between dBm0 and dBov. In that same section you can see that the G.711 A law, u law, and G.722 codec overload points are also defined.

Hi, on one of our clients asterisk server we have the problem that you hear nothing on external calls. Here are the details abount the system: Asterisk 1.6.0.22 DIGIUM Wildcard B410 quad-BRI card (rev 01) dahdi-linux-complete-2.2.0+2.2.0 I have setup the following test extension: exten => 9216992,1,ANSWER() exten => 9216992,2,WAIT(2) exten =>

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

A previous issue has popped up and once again I'm out of ideas. During the evenings it seems that the TDM channels will spike (dahdi_monitor) and will refuse to listen for audio of any type, this includes DTMF. The only resolution I know of is to stop Asterisk and restart the dahdi service, but that's not a solution. All channels look like this, even the FXS. [root at asterisk Hardware]#

Hi Marc, It's not exactly clear how to define the an "overload point" for Opus. In normal use, it's not a good idea to go beyond the +/- 32767 values for 16-bit PCM or +/- 1.0 for float. That being said, internally Opus can represent signals that are up to about 48 dB louder than the PCM saturation point and if you use the float implementation you can actually encode and decode

Hi, On one of our servers, we're having problems with incoming faxes. The connections come in through a PRI into a Digium TE820 card. 'fax show stats' yields the following: FAX Statistics: --------------- Current Sessions : 0 Reserved Sessions : 0 Transmit Attempts : 0 Receive Attempts : 740 Completed FAXes : 740 Failed FAXes : 486 Spandsp G.711

Hi, Using 1.4.26.1 & DAHDI 2.2.0.2, FSK VMWI devices off a TDM840 work fine. With 1.6.1.[45] & same DAHDI, instead of the FSK spill I get a line polarity reversal. Stutter dialtone is generated as expected. Has anyone else seen this? Is there anything special I need to do for 1.6.1 to make FSK MWI work? Thanks, --Barry

A few bits and pieces that I hope someone here has answers for. Basically, I'm in the audio business and I hope to get better control of the audio performance in *. 1. I seem to recall seeing a tonelist parameter that allowed tones to be set at levels other than 0, for instance !1000/500@-20 would generate 1 kHz for half a second at -20 dBm0. But this doesn't work and I can't

Hi Eric, > On 12 avr. 2016, at 17:48, Eric Cooper <ecc at cmu.edu> wrote: > > On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : >> [...] >> However, i can't find any dahdi startup script, neither init.d neither systemd

Now that I have a new card and my echo problems are 'mostly' solved, I have another major issue to deal with. After about an hour or so the card will stop detecting DTMF tones on incoming calls. dahdi_monitor shows the following: [root at asterisk wctdm24xxp]# dahdi_monitor 1 -v Visual Audio Levels. -------------------- Use chan_dahdi.conf file to adjust the gains if needed. ( # =

Hello, I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : $ sudo dpkg -l|grep -Ei 'dahdi|asterisk|libpri' ii asterisk 1:11.13.1~dfsg-2+b1 amd64 Open Source Private Branch Exchange (PBX) ii asterisk-config 1:11.13.1~dfsg-2 all Configuration

Hi, I want to add a new password hashing scheme as plugin and provide it for the dovecot project, so that it will be included as optional plugin in future releases. Yet the plugin compiles fine and the .so file gets created. My approach is to call the functions password_scheme_register() and password_scheme_unregister() (src/auth/password-scheme.c) inside the plugin's _init() and _deinit()

Hello again list, I'm running a 1.4.42 install on SUSE with an AEX410P card. The DAHDI release is 2.4.0 because the machine won't properly install 2.5 and also won't install Asterisk 10.0 because I can't get a good SQLite3 library to install. Whenever I enter DTMF very quickly or very slowly, app_read des on me. Has anyone experienced similar joy

2016-07-31 16:39 GMT+02:00 <aki.tuomi at dovecot.fi>: > > > On July 27, 2016 at 2:08 AM Andreas Meyer <luckyfellow42 at gmail.com> > wrote: > > > > > > Hi, > > > > > > I want to add a new password hashing scheme as plugin and provide it for > > the dovecot project, so that it will be included as optional plugin in > > future

Hi list, i'm using Samba 3.0.14a as standalone-pdc with posix-acl-features. On the fileserver i've mounted an ext3 partition to /home /dev/hdc2 /home type ext3 (rw,usrquota,acl) The following directory structur exists on /home /home +-/public_folder (exported per samba as \\fileserver\public ) | +-/a | +-/a_ann | | +-/anyfolder | | +- anyfiles | +-/a_bert | +-/a_sven | +-/a_eve |

> On August 1, 2016 at 4:38 PM aki.tuomi at dovecot.fi wrote: > > > > > On August 1, 2016 at 3:45 PM Andreas Meyer <luckyfellow42 at gmail.com> wrote: > > > > > > 2016-07-31 16:39 GMT+02:00 <aki.tuomi at dovecot.fi>: > > > > > > > > > On July 27, 2016 at 2:08 AM Andreas Meyer <luckyfellow42 at gmail.com> >

Hello I am working on TDM2400p. I am having some problems like: when i connect my analog phone with the card there is no dial tone, but i can dial any extension... but after that i can't hear any voice from my receiver i have used different phone sets but still i cant communicate with other extension. Please help me out. Thank you Regards Ali Raza -------------- next part -------------- An

Hi All; In dahdi: what we use instead of ztcfg -vv (that is existed /sbin/ztcfg -vv). ? Regards Bilal