subject:"\[Asterisk\-Users\] DTMF Error"

Re: [asterisk-users] DTMF error on asterisk

2007-09-13 Thread Eric "ManxPower" Wieling

HNAGUPCAUSE is more specific.  Cause 31 is "Normal, Unspecified" end of 
call.  Chances are it is a harmless message and is a telco caused issue.

See 
http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf

gincantalupo wrote:
> Hi satish,
> I get that error too (my Asterisk version is 1.2.x but should be the 
> same) when that Zap channel is not available and you are trying to use it.
> You should get a CHANUNAVAIL from Asterisk channel status.
> 
> Giorgio.
> 
> 
> satish patel wrote:
>> Dear all
>>
>>I have asterisk 1.4.11 on centos 4.x i have installed 2 
>> PRI on is asterisk and it is working fine but i got this DTMF error on 
>> asterisk CLI what is it ??
>>
>>
>> -- Zap/36-1 is ringing
>> -- Zap/36-1 answered SIP/5406-9fa59770
>> -- Channel 0/1, span 2 got hangup request, cause 31
>> [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: 
>> Unable to forward voice or dtmf

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Re: [asterisk-users] DTMF error on asterisk

2007-09-13 Thread gincantalupo

Hi satish,
I get that error too (my Asterisk version is 1.2.x but should be the 
same) when that Zap channel is not available and you are trying to use it.
You should get a CHANUNAVAIL from Asterisk channel status.

Giorgio.


satish patel wrote:
> Dear all
>
>I have asterisk 1.4.11 on centos 4.x i have installed 2 
> PRI on is asterisk and it is working fine but i got this DTMF error on 
> asterisk CLI what is it ??
>
>
> -- Zap/36-1 is ringing
> -- Zap/36-1 answered SIP/5406-9fa59770
> -- Channel 0/1, span 2 got hangup request, cause 31
> [Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: 
> Unable to forward voice or dtmf
> -- Hungup 'Zap/32-1'
>
>
>
>
>
>
> 
> Pinpoint customers 
> who
>  
> are looking for what you sell.
> 
>
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-- 

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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[asterisk-users] DTMF error on asterisk

2007-09-13 Thread satish patel

Dear all

   I have asterisk 1.4.11 on centos 4.x i have installed 2 PRI on 
is asterisk and it is working fine but i got this DTMF error on asterisk CLI 
what is it ?? 


-- Zap/36-1 is ringing
-- Zap/36-1 answered SIP/5406-9fa59770
-- Channel 0/1, span 2 got hangup request, cause 31
[Sep 13 22:10:29] WARNING[7191]: app_dial.c:741 wait_for_answer: Unable to 
forward voice or dtmf
-- Hungup 'Zap/32-1'







   
-
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RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

2004-01-29 Thread John Todd

I think what he was talking about was the fact that Grandstream 
phones send "f" as a DTMF signal when one hits the "flash" button.

JT

At 11:23 AM -0600 1/29/04, Mark Spencer wrote:
Latest CVS should not detect 'f' except in the case of a real fax.

Mark

On Thu, 29 Jan 2004, Brent Franks wrote:

 Christian,

 You can change channel.c source code to be more forgiving of
 unrecognized DTMF tones.
 Look for my addition near the bottom of this struct:

	else if (digit == 'f');

 Basically I altered channel.c to this:

 static int do_senddigit(struct ast_channel *chan, char digit)
 {
 int res = -1;
 if (chan->pvt->send_digit)
 res = chan->pvt->send_digit(chan, digit);
 if (!chan->pvt->send_digit || res) {
 /*
  * Device does not support DTMF tones, lets fake
  * it by doing our own generation. (PM2002)
  */
 static const char* dtmf_tones[] = {
 "!941+1336/50,!0/50",   /* 0 */
 "!697+1209/50,!0/50",   /* 1 */
 "!697+1336/50,!0/50",   /* 2 */
 "!697+1477/50,!0/50",   /* 3 */
 "!770+1209/50,!0/50",   /* 4 */
 "!770+1336/50,!0/50",   /* 5 */
 "!770+1477/50,!0/50",   /* 6 */
 "!852+1209/50,!0/50",   /* 7 */
 "!852+1336/50,!0/50",   /* 8 */
 "!852+1477/50,!0/50",   /* 9 */
 "!697+1633/50,!0/50",   /* A */
 "!770+1633/50,!0/50",   /* B */
 "!852+1633/50,!0/50",   /* C */
 "!941+1633/50,!0/50",   /* D */
 "!941+1209/50,!0/50",   /* * */
 "!941+1477/50,!0/50" }; /* # */
 if (digit >= '0' && digit <='9')
 ast_playtones_start(chan,0,dtmf_tones[digit-'0'], 0);
 else if (digit >= 'A' && digit <= 'D')
 ast_playtones_start(chan,0,dtmf_tones[digit-'A'+10], 0);
 else if (digit == '*')
 ast_playtones_start(chan,0,dtmf_tones[14], 0);
 else if (digit == '#')
 ast_playtones_start(chan,0,dtmf_tones[15], 0);
 else if (digit == 'f');
 else {
 /* not handled */
 ast_log(LOG_WARNING, "Unable to handle DTMF tone
 '%c' for '%s'\n", digit, chan->name);
 return -1;
 }
 }
 return 0;
 }
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Cristian
 Manoni
 Sent: Thursday, January 29, 2004 11:04 AM
 To: [EMAIL PROTECTED]
 > Subject: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for
 > SIP
 >
 > Hi All
 > i have continuos error:
 > Unable to handle DTMF tone 'f' for 'SIP
 > on the asterisk console.
 after this the call hang up.

 I have a BGT 101 that make and receive call from the capi channel

 Thanks
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RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

2004-01-29 Thread Mark Spencer

Latest CVS should not detect 'f' except in the case of a real fax.

Mark

On Thu, 29 Jan 2004, Brent Franks wrote:

> Christian,
>
> You can change channel.c source code to be more forgiving of
> unrecognized DTMF tones.
>
> Look for my addition near the bottom of this struct:
>
>   else if (digit == 'f');
>
> Basically I altered channel.c to this:
>
> static int do_senddigit(struct ast_channel *chan, char digit)
> {
> int res = -1;
>
> if (chan->pvt->send_digit)
> res = chan->pvt->send_digit(chan, digit);
> if (!chan->pvt->send_digit || res) {
> /*
>  * Device does not support DTMF tones, lets fake
>  * it by doing our own generation. (PM2002)
>  */
> static const char* dtmf_tones[] = {
> "!941+1336/50,!0/50",   /* 0 */
> "!697+1209/50,!0/50",   /* 1 */
> "!697+1336/50,!0/50",   /* 2 */
> "!697+1477/50,!0/50",   /* 3 */
> "!770+1209/50,!0/50",   /* 4 */
> "!770+1336/50,!0/50",   /* 5 */
> "!770+1477/50,!0/50",   /* 6 */
> "!852+1209/50,!0/50",   /* 7 */
> "!852+1336/50,!0/50",   /* 8 */
> "!852+1477/50,!0/50",   /* 9 */
> "!697+1633/50,!0/50",   /* A */
> "!770+1633/50,!0/50",   /* B */
> "!852+1633/50,!0/50",   /* C */
> "!941+1633/50,!0/50",   /* D */
> "!941+1209/50,!0/50",   /* * */
> "!941+1477/50,!0/50" }; /* # */
> if (digit >= '0' && digit <='9')
>
> ast_playtones_start(chan,0,dtmf_tones[digit-'0'], 0);
> else if (digit >= 'A' && digit <= 'D')
>
> ast_playtones_start(chan,0,dtmf_tones[digit-'A'+10], 0);
> else if (digit == '*')
> ast_playtones_start(chan,0,dtmf_tones[14], 0);
> else if (digit == '#')
> ast_playtones_start(chan,0,dtmf_tones[15], 0);
> else if (digit == 'f');
>     else {
> /* not handled */
> ast_log(LOG_WARNING, "Unable to handle DTMF tone
> '%c' for '%s'\n", digit, chan->name);
> return -1;
> }
> }
> return 0;
> }
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Cristian
> Manoni
> Sent: Thursday, January 29, 2004 11:04 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for
> SIP
>
> Hi All
> i have continuos error:
> Unable to handle DTMF tone 'f' for 'SIP
> on the asterisk console.
> after this the call hang up.
>
> I have a BGT 101 that make and receive call from the capi channel
>
> Thanks
> ___
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RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

2004-01-29 Thread Brent Franks

Christian,

You can change channel.c source code to be more forgiving of
unrecognized DTMF tones.

Look for my addition near the bottom of this struct:

else if (digit == 'f');

Basically I altered channel.c to this:

static int do_senddigit(struct ast_channel *chan, char digit)
{
int res = -1;

if (chan->pvt->send_digit)
res = chan->pvt->send_digit(chan, digit);
if (!chan->pvt->send_digit || res) {
/*
 * Device does not support DTMF tones, lets fake
 * it by doing our own generation. (PM2002)
 */
static const char* dtmf_tones[] = {
"!941+1336/50,!0/50",   /* 0 */
"!697+1209/50,!0/50",   /* 1 */
"!697+1336/50,!0/50",   /* 2 */
"!697+1477/50,!0/50",   /* 3 */
"!770+1209/50,!0/50",   /* 4 */
"!770+1336/50,!0/50",   /* 5 */
"!770+1477/50,!0/50",   /* 6 */
"!852+1209/50,!0/50",   /* 7 */
"!852+1336/50,!0/50",   /* 8 */
"!852+1477/50,!0/50",   /* 9 */
"!697+1633/50,!0/50",   /* A */
"!770+1633/50,!0/50",   /* B */
"!852+1633/50,!0/50",   /* C */
"!941+1633/50,!0/50",   /* D */
"!941+1209/50,!0/50",   /* * */
"!941+1477/50,!0/50" }; /* # */
if (digit >= '0' && digit <='9')
 
ast_playtones_start(chan,0,dtmf_tones[digit-'0'], 0);
else if (digit >= 'A' && digit <= 'D')
 
ast_playtones_start(chan,0,dtmf_tones[digit-'A'+10], 0);
else if (digit == '*')
ast_playtones_start(chan,0,dtmf_tones[14], 0);
else if (digit == '#')
ast_playtones_start(chan,0,dtmf_tones[15], 0);
else if (digit == 'f');
else {
/* not handled */
ast_log(LOG_WARNING, "Unable to handle DTMF tone
'%c' for '%s'\n", digit, chan->name);
return -1;
}
}
return 0;
}

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cristian
Manoni
Sent: Thursday, January 29, 2004 11:04 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for
SIP

Hi All
i have continuos error:
Unable to handle DTMF tone 'f' for 'SIP
on the asterisk console.
after this the call hang up.

I have a BGT 101 that make and receive call from the capi channel

Thanks
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Re: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

2004-01-29 Thread Walter Doerr

On Thu, Jan 29, 2004 at 05:04:22PM +0100, Cristian Manoni wrote:
> Hi All
> i have continuos error:
> Unable to handle DTMF tone 'f' for 'SIP
> on the asterisk console.
> after this the call hang up.

Look at softdtmf in capi.conf.
Setting the parameter to 0 solved the problem for me.

-Walter



-- 
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  "The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck."  (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)
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[Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

2004-01-29 Thread Cristian Manoni

Hi All
i have continuos error:
Unable to handle DTMF tone 'f' for 'SIP
on the asterisk console.
after this the call hang up.

I have a BGT 101 that make and receive call from the capi channel

Thanks
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[Asterisk-Users] DTMF Error

2003-12-28 Thread Brent Franks

Hello,

On the Polycom IP 500 Phones, when I press the mic mute button, the mic
on the speaker or headset goes muted.  However when I press the mic mute
button again, the call is terminated by asterisk.  Asterisk shows a:

WARNING[1236268096]: File channel.c, Line 1296 (do_senddigit): Unable to
handle DTMF tone 'f' for 'SIP/-'

I am using reinvite=no on the phones.

After looking through the C file, it is clear where DTMF tones are
declared, however, I want to make this one silent.  Is there anyway I
could alter this file, to simply ignore the tone and continue the call.

Also, from a design perspective, should Asterisk natively be more
lenient with tones that it doesn't understand, rather than just cutting
the call off?  Not sure if this would be considered a bug or whether
there was some functionality with this that I just am missing?

Thanks,

Brent



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Re: [asterisk-users] DTMF error on asterisk

Re: [asterisk-users] DTMF error on asterisk

[asterisk-users] DTMF error on asterisk

RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

RE: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

Re: [Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

[Asterisk-Users] DTMF Error: Unable to handle DTMF tone 'f' for SIP

[Asterisk-Users] DTMF Error

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