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
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
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
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
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
--
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
;
}
}
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
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
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
