similar to: Open ITU G.107 Implementation to measure voice quality

Hi, I created a script named example2.sh which goal is read some text from my HP Service Desk using an application in java and send this text to the text2wave application for TTS. example2.sh java -Xbatch Example10 | text2wave -f 8000 -o /var/lib/asterisk/sounds/my-sd.wav When I execute the script in prompt, everything is ok, but when I use the system() command in my extensions.conf it isn?t

All, Having problems terminating to a Nuvox PRI, the tech at Nuvox is saying Asterisk is transmitting in CCITT (aka ITU?) when they're expecting (and will only accept) ANSI. The question is, is there a simple way to change this or am I stuck with rewriting code? I googled and checked the mailing list and found nothing, I could be barking up the wrong tree I guess. PRI is not my forte.

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

Hi, There is a lot of speex quality estimations. One of this comparative estimation is even available on the official site <http://speex.org/comparison/>. I'd like to present yet another one. And I thought that the best place for this presentation would be Speex-dev mailing list. I want to get feedbacks and criticisms please. If Speex authors consider to make some parts of this work

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Hi, I?m trying to make * work with Brazilian analog signalling.. I?m using the following in indications.conf file... [br] description = Brasil ringcadence = 1000,4000 dial = 425 busy = 425/250,0/250 ring = 425/1000,0/4000 callwaiting = 425/60,0/250,425/60,0/5000 I changed zaptel.conf to loadzone=br #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

When trying to make outbound calls I am getting the Warning: File app_dial.c line 313 (wait_for_answer) Unable to forward voice. When making the call it attempts to dial (pounds are actually numbers but replaced to not show numbers we are dialing): Executing Dial("Sip/donas-bd7b", Zap/g1/1##########") in new stack Called g1/1########## Channel 1, span 1 got hangup **Above

Hello, I am relatively new to Asterisk and I need to compile the G.723.1 codec for Asterisk. I downloaded the ITU source code, placed it in the codecs directory, but apparently Asterisk needs a rather different library than the one provided from ITU. As I've seen in the mailing list archives, there are quite a few users who were able to compile G.723.1 in *, so, could someone kindly share it

Does anyone have the C reference code of the ITU G.711 Appendix II ? -- Guilherme Loch G?es "Wave after wave will flow with the tide And bury the world as it does Tide after tide will flow and recede Leaving life to go on as it was..." - Neil Peart , Natural Science

Greetings list, Does anyone have any information (providers' support) about G.729E? Voip-info.org came up empty, the implementers guide from the ITU wants my credit card and the rest of the pages I found simply made a few comparisons between it and iLBC. >From what I understand, the codec is supposed to play nicely on lower power hardware but I can't find much more info than that.

Hi, The ITU has been churning out a series of codecs which stretch older codecs to support wideband or super wideband and stereo. Does anyone know of these things being used in the real world? They push the idea of these things offering high compatibility, as the narrowband bitstream is embedded in the wideband stream. However, I'm skeptical this offers any real world advantage.

Hi, I have both codec_g726.so and format_g726.so loaded: root at test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I

hi; after i've compiled Samba3 with supports(ads,krb5,acl,winbindd) and have configured smb.conf having workgroup = CC realm = CC.AD.ITU.EDU.TR security = ADS idmap uid = 500-65535 idmap gid = 100-65535 winbind separator = + winbind cache time = 15 krb5.conf having [realms] CC.AD.ITU.EDU.TR = { kdc = atreides.cc.ad.itu.edu.tr:88 admin_server = atreides.cc.ad.itu.edu.tr:749

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990

On 7/16/15, Martin Leese <martin.leese at stanfordalumni.org> wrote: > Martijn van Beurden wrote: >> I would propose: 0000-0111 : (number of independent channels)-1. >> The channel order is defined through the >> WAVEFORMATEXTENSIBLE_CHANNEL_MASK vorbis comment, if defined. If >> no WAVEFORMATEXTENSIBLE_CHANNEL_MASK is present, the channel >> order follows

Hello, I need to convert the dahdi_monitor output to dBm0, so I can measure Echo Return Loss in dB. I've read a formula that calculate S(k) in ITU-T G.168 recommendation, where S(k) is the signal level in dBm0. Can I use this formula to convert it? If yes, what value should I use to the number of samples if I want to convert a single output from dahdi_monitor? -- Thanks in advance, Gustavo