search for: voice

......) } subset(m1data,! is.na(votms) & !is.na(age) & !is.na(Cplace) & !is.na(tokentype)& !is.na(voicing), select=c(votms,age,Cplace,tokentype,voicing)) -> sampledata xyplot(votms ~ age | Cplace + tokentype,data=datasample,xlab="Subject Age (months)",ylab="Voice Onset Time (ms)",panel="my.panelconstr",groups=voicing) Some information on the data set: > sapply(sampledata,levels) > $votms > NULL > > $age > NULL > > $Cplace > [1] "bilabial" "dental" "velar" > > $token...

Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? Thank you, Pim

...mestamps log datetime msec no service password-encryption ! hostname Asterisk_IAD2430 ! boot-start-marker boot system slot0:c2430-i6s-mz.123-4.T7.bin boot-end-marker ! enable password asterisk26 ! username asterisk no aaa new-model ip subnet-zero no ip domain lookup ! ! no ftp-server write-enable ! voice call send-alert voice rtp send-recv ! voice service voip sip ! voice class codec 1 codec preference 1 g711alaw codec preference 2 g711ulaw codec preference 3 gsmfr ! ! ! ! ! ! ! ! ! ! ! ! interface FastEthernet0/0 ip address 192.168.0.206 255.255.255.0 speed 100 full-duplex ! interface Fast...

...to the asterisk sip server and I see the name I statically assign in the Cisco appears on the terminating end (linksys ata) I use the command in the Cisco under sip-ua calling-info pstn-to-sip from name set name timers buffer-invite 5000 I have also tried to add the commands... remote party-id -voice service voip,sip ds0-num Basically I need to take the field Remote-party id and place it in the sip message "From" Here is some debug from the sip messages in the Cisco... this is an example of "no caller id Name" Sent: INFO sip:5132017005@65.23.9.xxx:5060 SIP/2.0 Via: SIP/...

...0600, Kristian Kielhofner wrote: > > > >> We are getting pricing and one provider is telling us > that they have > >>quality issues with the "Integrated Access" product. From > what they > >>say it sounds like you can have audio dropouts on the voice > channels > >>when the data channels are being pushed to the max. I have never > >>heard of this. It doesn't seem possible to me, and if at all, > >>probably more of a limitation in the CSU/DSU than in the actual T. > >>Isn't that what TDM is...

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

Hi All, I have some Polycom IP300's and I'm interested in increasing the max volume for the headset (not handset), I'm wondering if anyone has experience adjusting these values: <gains voice.gain.rx.analog.handset="0" voice.gain.rx.analog.headset="0" voice.gain.rx.analog.chassis="3" voice.gain.rx.analog.chassis.obs="-12" voice.gain.rx.analog.chassis.IP300="-6" voice.gain.rx.analog.ringer="3" voice.gain.rx.analog.ringer.IP300=&...

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote: > On 4/5/19 10:36 AM, sean darcy wrote: > > I'm trying to set up pjsip to work with an obi202 and google voice. But > > I can't configure the endpoint. > > > > pjsip: > > > > [obi202-auth](!) > > type = auth > > auth_type = userpass > > password = <mypass> > > > > [obi202-aor](!) > > type = aor > > max_contacts = 2 > &g...

...gs: Cisco ------ ! version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname 192.168.65.200 ! logging queue-limit 100 enable secret enable password ! ip subnet-zero ! ! no ip domain lookup ! isdn switch-type primary-net5 ! ! voice call carrier capacity active ! voice service pots ! voice service voip ! voice class codec 10 codec preference 1 gsmfr codec preference 2 g711alaw ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! ! mta receive maximum-recipients 0 ! ! controller T1 1/0 shutdown fr...

I'm trying to set up pjsip to work with an obi202 and google voice. But I can't configure the endpoint. pjsip: [obi202-auth](!) type = auth auth_type = userpass password = <mypass> [obi202-aor](!) type = aor max_contacts = 2 ; ===== endpoints ======== [gv-voice](obi202-endpoint) auth = gv-voice aors = gv-voice identify_by=auth_username ;identify_b...

Hello, I am currently pricing out various T1 and PRI options for a client of mine. We need voice and data - we want T's. Whether it be two seperate T's, two superate fractional T's, or one combined fractional T, we need it done. We are getting pricing and one provider is telling us that they have quality issues with the "Integrated Access" product. From what they s...

...tting the chan_capi module to load in asterisk cvs from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva Server Bri card. I load the modules with: modprobe -v divas divacapi I load the firmware with: divactrl load -c 1 -f ETSI -vd6 Output in /var/log/messages is: Nov 9 19:26:26 voice kernel: Eicon DIVA - DIDD table (http://www.melware.net) Nov 9 19:26:26 voice kernel: divadidd: Rel:2.0 Rev:1.13 Build:102-51(local) Nov 9 19:26:26 voice kernel: Eicon DIVA Server driver (http://www.melware.net) Nov 9 19:26:26 voice kernel: divas: Rel:2.0 Rev:1.45 Build: 102-52(local) Nov 9...

Can someone explain me this message: "chan_iax2.c: Ooh, voice format changed to ..." Where can I find a list of numeric codes used to identify voice format? Then, sometime I get an infinite loop of messages like these: DEBUG[15015] chan_iax2.c: Ooh, voice format changed to 1 WARNING[15015] channel.c: Unable to find a codec translation path from g723...

...ort FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to call F, and it will ring. Voice from softphone to F carries over and I can hear it; however, no voice from F to softphone will carry. I have been experimenting with different codec and other cisco/asterisk config tips from the web. None had worked so far. If anyone have experienced such problem and knows how to solve this, I wil...

...queue that will handle inbound calls to dynamically added agents that are all mobile numbers. Now when I do this setup it works, it loads the agents dynamically and if the mobile phone is on and have reception it works. But when the phone is for arguments sake off or dont have reception it goes to voice mail for that mobile phone. I don't want this to happen...:) I would like for the queue to continue ringing until there is a time out specified which then takes the caller out of the queue and to voice mail which I then intend to mail somewhere. I guess my question is can this be done in Aste...

I have a asterisk server. Two SIP Soft XLites are connected to the server. I am able to make calls from one SIP Phones to the other SIP Phones and landlines successfully. The SIP Soft Phone on th eother side can hear my voice but I cannot hear their voice. They can call my local cell phone as well. Samething, they can hears my voice, I cannot hear their voice. The microphone and speakers are working on both sides because we are able to use google talk and are able to talk successfully. But it would not work on XLite...

Hi: i have a problem. Mi extensions.conf: exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN}) exten => _N.,2,SetAccount(${customer}) exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1}) exten => _N.,4,ResponseTimeout(5) exten => _N.,5,Background(ifyou) exten => _N.,6,Background(silence/1) exten => _N.,7,Background(ifyou) exten => _N.,8,Background(si...

?I have connection with two networks (by VoIP provider setup) 1 - 10.10.10.0/24 = SIP 2 - 10.10.11.0/24 = Voice How to tell Asterisk send / receive voice traffic not on SIP network. When I look into dumps, I see Asterisk trying to use SIP net for voice Unfortunately, I _need_ to use two networks instead of one? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists...

...latest chan_capi (also tried an older version). When using the Capi-Channel, everything works fine except from the sound it sounds extremely choppy and is unusable :-( When e.g. capisuite is used for fax, everything sounds fine... I found the following when using capi debug: ISDN1#02: too much voice to send for NCCI=0x10101 Google finds nothing relevant for this error message :-( Has anybody any idea ? Christoph P.S.: Here is the output of capi debug CONNECT_IND ID=002 #0x016e LEN=0037 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10 CalledPartyNumber...

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play a beep. And then we record his voice and realize voice recognition with ispeech...