similar to: Requested transfer capability: 0x00 - SPEECH - How to change to 31KAUDIO?

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hi, on a configuration with one external ISDN S bus (to telco) and one internal S bus (to ISDN telephone), where Asterisk is in the middle (using HFC hardware), I noted the following: - when a GSM phone or ISDN phone calls in, the Transfer capability is Requested transfer capability: 0x00 - SPEECH - when an analog phone calls in (either from an analog line or an analog ISDN

Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: *zaptel.conf:* span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be *zapata.conf:* [trunkgroups]

I used Microsoft text to speech with speex. It sounded reasonably well. I didn't experiment with various quality settings (8KHz, default quality). I was capturing audio from a sound card, not the direct wave data from the TTS engine. .a -----Original Message----- From: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] On Behalf Of Reed Hedges Sent: Friday, May 26, 2006 7:23 AM

I am a newbie here so this may be a well worn topic... I am working on a cd compilation that would combine speech such as lectures and music. I have many speeches that are 60 to 90 minutes in length and would like to compress them as much as possible. I have one in mp3 format with the following specs: Size: 13603893 bytes Header found at: 0 bytes Length: 3733 seconds MPEG 2.5 layer 3 29kbit

Hello, I just installed a PRI and when I make a local (seven digit) call, I get Code 28 back from the telco, (I believe code 28 means "Invalid Number") and I hear a fast busy on the phone. Here is the output: -- Executing Dial("SIP/marke-17b1", "ZAP/G1/4967171") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called G1/4967171

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it as-is with the dahdi one ; to no

Tom Grandgent wrote: > Have you tried using 16kHz wideband? The sound quality is far superior to > narrowband, IMO, even if you have to turn the VBR quality down (to say, 2) > to save bandwidth. Thanks for the info Tom! Probably narrowband is hurting me, but my system is currently built on that. I want to try to get acceptable performance from narrowband if possible before trying

Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as

Only a warning, and doesn't seem to do anything bad. But I can't seem to figure out what these warnings mean? -- Requested transfer capability: 0x00 - SPEECH [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: k [Feb 17 12:33:03] WARNING[10750]: chan_dahdi.c:3096 dahdi_call: Unrecognized prilocaldialplan NPI modifier: o [Feb 17

Briefly: I want to be able to have "press or say (number)", with Asterisk listening for a spoken number, but accepting a DTMF digit, too. I'm posting everything I found so far, here, partly to show working, but also in case anyone else finds it useful. So, moving on.... This looked hopeful for a moment until I realised that it doesn't do DTMF:

This week Steve Sokol stops by to describe and field questions about Digium's new affordable speech recognition solution. Later on in the call, we'll also be looking at iVoIP, clients and uses for mobile VoIP. Join us on IRC anytime #voip-users-conference During the conference, call via SIP g711 or wideband g722 - or Try the web page widget to call in wideband. The details on all the

Hi Reed, I've been using Speex to transmit TTS for years. It works very well with no tweaking. I use Microsoft TTS ("Microsoft Mike") with Speex at 16kHz wideband and VBR quality 6. Sometimes I forget that the sound is even coming from another computer and being compressed+decompressed. If anything, TTS seems easier for Speex to deal with than real voice. But I don't

Hi, I'm having strange warning from asterisk when I try to dial GSM Gateway: -- Executing [1011501522xxx at sm:1] NoCDR("SIP/ibm-b2c52848", "") in new stack -- Executing [1011501522xxx at gsm:2] Dial("SIP/ibm-b2c52848", "Zap/R3/501522xxx") in new stack -- Requested transfer capability: 0x00 - SPEECH [Jul 13 11:58:50] WARNING[18208]:

Hello, I'm wondering if anyone has done this before and has any advice, or if anyone in general has ideas about it. I just implemented transmitting synthesized speech (text-to-speech) over Speex (narrowband) in an application. I'm using Swift from Cepstral (http://www.cepstral.com). The voice I'm using is a pretty deep male voice. I'm telling Swift to generate audio at 8khz,

I guess it is possible to have an HFC-S card and a Digium TE110P card working at the same time? The TE110P will work in E1 mode. I think the zaptel.conf is probably right but the zapata.conf not (I just tacked on another group at the end but I don't really know what I am doing). Can anyone help? ----------------------------------------- My zaptel.conf looks like this: # hfc-s pci a span

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

I've got a bit trouble with callerid and zaphfc cards. Basically zaphfc doesn't add the "0" in front of national numbers (haven't tried a international call yet). With chan_capi that allways worked fine, however i had to define the national and international prefixes in capi.conf. Is there something similar in zapata.conf ? Here is my zapata.conf: [channels]

Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown