similar to: RE: Grandstream BT-100 and

Hello list! I'm using "Asterisk CVS-11/22/03-04:28:51" and try to route my normal (classic) phone calls via nikotel (www.nikotel.com). I can talk about 1 minute and get then disconnected. Here my current configuration parts which affect nikotel: register => chabrol:PASSWORD_REMOVED@nikotel/500 [nikotel] type=friend secret=PASSWORD_REMOVED username=chabrol fromuser=chabrol

Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a "front end" for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or

My setup is as follows: Handset -> Sipura SPA 2000 -> Asterisk -> VoicePulse and Handset -> Sipura SPA 2000 -> Asterisk -> Digium X100P -> POTS I notice when making VoicePulse calls (but *not* POTS calls through the X100P) that there is significant "popping" and "clicking" on the line. This isn't enough to interfere seriously with the call, and

Hi, The problem the other way around: IceS -> IceCast -> ShoutCast -> KiSS (the KiSS then decodes the mp3 to a RAW SPDIF stream, which is received by my sourround receiver). But the constallation is somewhat overkill and slightly unstable, so I would prefer just IceS -> IceCast -> KiSS But that does not work at all for unknown reasons (since shoutcast can do this, and the

I have successfully set up a system with Ices0.4 and IceCast2.20 (and Tunez). It plays well using mpg123 or WinAmp. But I would like to use also my KiSS DVD player for the stream, and I have not been able to do so - It appearently needs 110% SHOUTcast compatibility. So I tried the IceCast2.3.1 - but it still didn't like the KiSS (or the other way around). In stead, I have added a SHOUTcast

Hello list! I only can make successful calls if I disable gsm with "disallow=gsm". As soon as I allow gsm the following appears at the console. There are much much more Lines with "File dsp.c, Line 1198" but I cut them for a better survey : --------- Log Start ------------- Asterisk Ready. WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on

I have been hunting around the inet trying to find a CLI BT tracker/hoster. I want to use one of my linux servers as a BT torrent hoster/tracker. Anyone have any good informaiton they can point me to? -- My "Foundation" verse: Isa 54:17 No weapon that is formed against thee shall prosper; and every tongue that shall rise against thee in judgment thou shalt condemn. This is the

Hi, I am trying to install and make functiona PC Navigator 8 for my father on Ubuntu 9.10 using wine. Software is runnable, i connect BT GPS receiver to ubuntu on rfcomm4, but I am unable to make receiver visible for PC Navigator. Can anyone help he? Thank you

Hello, I'm trying to connect a Digium Wildcard TE110P T1/E1 Card to a 10 channel BT ISDN 30 box, however I'm fairly new to Asterisk and this is the first time I've set-up a telephony interface card can anyone give me some pointers or known working configurations? I'm running Asterisk 1.6 and Dahdi 2.4.1.1. Thanks, Tanuki -------------- next part -------------- An HTML attachment

I have patches to add locking to the adv(4), adw(4), aha(4), ahb(4), aic(4), and bt(4) drivers so that they no longer use Giant. I would appreciate any testing folks are able to do. The patches are against HEAD but should apply to 8 or 9. Make sure you have INVARIANTS and INVARIANT_SUPPORT enabled for any initial testing that you do. Thanks!

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On Jul 9, 2014 5:15 AM, "Michael D. Setzer II" <mikes at kuentos.guam.net> wrote: > Subject: isohybrid binary not in 6.03-pre18? (utils bt is in bios/utils?) Correct, it's in the bios/ architecture binary directory. 6.00 introduced efi32/ and efi64/ making the distinction of the binaries necessary. --Gene

On Jul 9, 2014 6:11 AM, "Michael D. Setzer II" <mikes at kuentos.guam.net> wrote: > > On 9 Jul 2014 at 6:01, Gene Cumm wrote: > > Date sent: Wed, 9 Jul 2014 06:01:22 -0400 > Subject: Re: [syslinux] isohybrid binary not in 6.03-pre18? (utils bt is in bios/utils?) > From: Gene Cumm <gene.cumm at gmail.com> > To: "Michael D. Setzer,

> > But these is no isohybrid in the efi directories? So, why wasn't the isohybrid.c > also moved to bios/utils? The source code is still in the same place; the binaries are generated into respective subdirectories. I would suggest reading: http://www.syslinux.org/wiki/index.php/Library_modules http://www.syslinux.org/wiki/index.php/Common_Problems#Multi-Arch and in the

Hello I was wondering whether anyone has managed to get an X100P card to receive Caller ID information in the UK (connected to a BT line)? I guess that it probably won't work at present because BT sends the CID information before the first ring - hence there is no event to wake up the channel. I can hear it using ztmonitor, but it is far too soon for anything to receive it. According to the

Hi Folks, If anyone is interested I now have a BT Easicom 1000 working with *. Some initial problems but they are sorted. Everything seems to work ok with the exception of the vm script. At the moment I basically have to decline the download for it to work.. The really good news is that I guess these phones aren't locked. Nice big screen and there's a pullout keyboard too :D Andy

Hello, I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Regards, Christopher

Have anyone tried to interface BT's Broadband Voice with asterisk? Kannaiyan

Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119&category=main

Hi, When we use BudgeTone-100 in our Intranet together with our Asterisk IP PBX everything is working OK. When we try to use the phone behind the Firewall we can't do the connection. When I try to use STUN Server: 128.107.250.38 there is no result. The only way in which I have audio from the one direction (BT-100 to Asterisk) is when I leave blank STUN Server and specify the IP Address in