search for: voices

I appologize if this post is a duplicate, but I did not get my first post myself.. Here is my original post: --------------------------------- Hi all, I've got some troubles with a Lattice plot. The error I get is Error in Ops.factor(groups[subscripts], vals[i]) : Level sets of factors are different which sort of doesn't make sense to me. Can you please help me figure out what

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

Hello, I have a cisco ATA 188 registering both of its lines to * I can place calls between then an to kphone an MSN messenger (both registering with * too), a few days ago a friend lend me a Cisco IAD 2430 and I was willing to do the same thing with it, since it has 24 ports I was willing to to use 24 analog phones with it however something really weird happens I can place calls from my ata,

All, I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is.... One pri terminating into a Cisco 2431 router Sip messages from the Cisco get sent to a asterisk server linksys ata's a each remote end. I can receive the calling name if the call originates

> Mark Farver wrote: > > > On Fri, 2004-12-17 at 16:26 -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 >

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"

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> > > >

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

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_by=username ; I also tried

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

Hello, I have an issue getting 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

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 to alaw DEBUG[15015]

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port 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

Hi All, Lets say I want to setup a 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

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

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(silence/5) exten

?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:

Hi everybody, I have a problem which I cannot eliminate on my own. Has anybody any idea for the following: I am using the asterisk-version from Debian-Testing (1.2.13) with the 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

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