RE: [Asterisk-Users] Intresting.. hrm

2003-08-23 Thread Brian West
OUCH.. its a nice phone but really is it that nice.. it runs linux so I wonder how hard custom firmware would be? bkw On Sat, 23 Aug 2003, Andrew Joakimsen wrote: Linuxdevices says $400 http://www.linuxdevices.com/articles/AT9406437906.html -Original Message- From: [EMAIL

[Asterisk-Users] Intresting Vonage story...

2003-08-23 Thread Brian West
http://www.politechbot.com/p-05040.html Funny... bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Intresting.. hrm

2003-08-23 Thread Brian West
But it says it includes power and headset also.. not bad. bkw On Sat, 23 Aug 2003, Andrew Joakimsen wrote: Linuxdevices says $400 http://www.linuxdevices.com/articles/AT9406437906.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling

Re: [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)

2003-08-23 Thread Ing. Angel Gomez Garcia
mhhh, Tan Aks wrote: For the FXS unit: 1) it doesn't recognise voicemail waiting messages, so your analog phones won't receive a stuttered dial tone. Right 2) it doesn't seem to recognise the transfer (#) button since it seems to use different payload numbers (rtp codec 100 and 96).

[Asterisk-Users] Vonage ATA-186 password recovery

2003-08-23 Thread Scrotus Maximus
This message describes the configuration and recovery process for Cisco ATA-186 adapters provided by Vonage. Every Vonage Customer Gets a Cisco Phone Adapter for Free. The unadvertised detail is that this adapter is never under your control, even after completing the terms of your customer

Re: [Asterisk-Users] Vonage ATA-186 password recovery

2003-08-23 Thread Scrotus Maximus
#include stdlib.h #include string.h #include stdio.h #define ATA_MAGIC #ata unsigned char bcd_lookup[100] = { 0x0, 0x1, 0x2, 0x3, 0x4, 0x5, 0x6, 0x7, 0x8, 0x9, 0x10, 0x11, 0x12, 0x13, 0x14, 0x15, 0x16, 0x17, 0x18, 0x19, 0x20, 0x21, 0x22, 0x23, 0x24, 0x25, 0x26, 0x27, 0x28, 0x29,

[Asterisk-Users] callerid, callwaiting callerid, Asterisk and ATA

2003-08-23 Thread Dan
Hi, There is any difference in the way callerid is transmitted when a call is received (in callwaiting mode- when in a conversation) or a regular call (phone on-hook)? I have a phone connected to ATA which receive and display the callerid during callwaiting, but not when the phone is on hook.

Qwest CallerID question re:[Asterisk-Users] Caller ID problem

2003-08-23 Thread John Brown
So after spending about 45 min with several completely clueless qwest people I ask here: ;) Does qwest provision callerid with DTMF or FSK ?? Is there a way to (using butt set or similar) to tell which method is used ? On Sat, Aug 23, 2003 at 03:23:49PM +0300, Dan wrote: Hi, First thing

Re: Qwest CallerID question re:[Asterisk-Users] Caller ID problem

2003-08-23 Thread Ryan Tucker
On Sat, 23 Aug 2003, John Brown wrote: So after spending about 45 min with several completely clueless qwest people I ask here: ;) Does qwest provision callerid with DTMF or FSK ?? Is there a way to (using butt set or similar) to tell which method is used ? Assuming you're in the United

Re: Qwest CallerID question re:[Asterisk-Users] Caller ID problem

2003-08-23 Thread John Brown
Yup, US Based, POTS directly from ILEC (Qwest, USWest, MountainBell) Ok, so either DTMF or data burst.. got it. THanks, off to telco room to play :) On Sat, Aug 23, 2003 at 01:33:10PM -0400, Ryan Tucker wrote: On Sat, 23 Aug 2003, John Brown wrote: So after spending about 45 min with

[Asterisk-Users] One-way audio using console

2003-08-23 Thread Jan Rychter
I've tried making calls using the console (both ALSA and OSS). ALSA seems to work after applying the little fix posted on this list some time ago by someone (which I'll submit into the bug tracker), but all I get is one-way audio: I can hear the other end, but nothing gets transmitted. At first I

Re: [Asterisk-Users] Intresting.. hrm

2003-08-23 Thread mawali
I have seen it in Linuxworld t ostel's booth. It is actually very close to (but bigger) the design I suggested in the list a couple of days ago. Beside long boot time a and huge memory, it seems to be running fine. The huge memory requirement is probably what drives the cost this high. Regards

RE: [Asterisk-Users] Intresting.. hrm

2003-08-23 Thread mawali
Linux ensures that custom firmware is a piece of cake. It actually makes pulling one feature out and putting another in as easy as on you computer. It is matter of adding a .so file and an executable to your filesystem. Switching stacks is as easy as killing one process and starting another. I

[Asterisk-Users] zaptel compile problems

2003-08-23 Thread John Brown
Hi list, I'm having problems getting zaptel to compile. I'm not a big Linux person and so don't know all the nifty ways RH does things. If this was FreeBSD it wouldn't be an issue :) here is the first few lines from the make gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__

Re: [Asterisk-Users] zaptel compile problems

2003-08-23 Thread mawali
Im sorry, but I would try dumping RedHat kernel. I have never been able to compile any kernel code with distro kernels. Use Vanilla kernel from ftp.kernel.org FT On Sat, 23 Aug 2003, John Brown wrote: Hi list, I'm having problems getting zaptel to compile. I'm not a big Linux person

RE: [Asterisk-Users] zaptel compile problems

2003-08-23 Thread Scott Stingel
All four sets (asterisk, libpri, zaptel, and zapata) compile fine under redHat 9.0 (although with lots of warnings) Looking at your /usr/src listing - shouldn't there be separate directories for asterisk, libpri, zaptel, and zapata? Have you checked that you have all the required packages

Re: Qwest CallerID question re:[Asterisk-Users] Caller ID problem

2003-08-23 Thread James Sharp
On my SBC phone, I used to hear a high-pitched chirp before the Call Waiting beep (much like the first chrip of a V.90 modem negotiation tone) when someone called in and I was on the line. Does this mean SBC was using FSK to transmit caller ID on my line? Yup. That's CallerID over Call

Re: [Asterisk-Users] There is any cache for sound files?

2003-08-23 Thread Dan
Finaly I have restarted the computer. Vry strange still the old prompts. Just a single example: demo-echotest.gsm I have defined an extension with the only line: exten = 111,1,Playback(demo-echotest) The file demo-echotest.gsm is changed (localized). When I dial this extension I still

Re: [Asterisk-Users] SIP change...

2003-08-23 Thread Mark Spencer
Normally the caller-id is taken from remote-party-id in the SIP INVITE. We don't see that field poplated in this INVITE. What is the originating gateway? What device is sending the call to the 827? We should be seeing remote-party-id in the INVITE. The string remote-party-id does not even