Re: [Asterisk-Users] newb question regarding DTMF
On Tue, 24 Aug 2004, Erik Anderson wrote:
> On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
> <[EMAIL PROTECTED]> wrote:
> > x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> > transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> > for your x-lite user.
>
> That's what I've read, and I have added dtmfmode=rfc2833 in my
> sip.conf...see this snippet:
>
> [xlite1]
> ;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
> ;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
> type=friend
> username=xlite
> callerid="Jane Smith" <5678>
> host=dynamic
> nat=yes ; X-Lite is behind a NAT router
> canreinvite=no; Typically set to NO if behind NAT
> disallow=all
> allow=gsm ; GSM consumes far less bandwidth than ulaw
> allow=ulaw
> allow=alaw
> dtmfmode=rfc2833
>
> I've applied that change and restarted asterisk, but no dice...
Dial the extension, then on the * CLI use 'sip show channels' to get the
name of the active channel. Next use 'sip show channel ___' to get info on
that particular channel (you can type the first few characters and use tab
completion; no need to type the whole string!). Scan through the output to
see whether asterisk is really using rfc2833 for that channel. If it is,
then the problem is likely in the x-lite config. If not, try moving
dtmfmode to the general section of sip.conf
Greg
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Re: [Asterisk-Users] newb question regarding DTMF
On Tue, 24 Aug 2004 11:46:36 -0600 (MDT), Greg Hill
<[EMAIL PROTECTED]> wrote:
> x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to
> transmit inband). You need dtmfmode=rfc2833 in [general] or in the section
> for your x-lite user.
That's what I've read, and I have added dtmfmode=rfc2833 in my
sip.conf...see this snippet:
[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=xlite
callerid="Jane Smith" <5678>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
dtmfmode=rfc2833
I've applied that change and restarted asterisk, but no dice...
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Re: [Asterisk-Users] newb question regarding DTMF
On Mon, 23 Aug 2004, Erik Anderson wrote: > Hello all - I'm just starting to play around w/ asterisk, and I've run > into a seemingly simple problem that has really manged to frustrate > me... > > I'm running the latest cvs version of *, and am trying to dial in to > the default extention 1000 demo using x-lite. I can dial and hear the > greeting no problem, but when I try and send any DTMF tones, I don't > get any response. Is there something specific I need to set in my > sip.conf to allow DTMF? x-lite uses the RFC2833 style for DTMF "out of the box" (it can be set to transmit inband). You need dtmfmode=rfc2833 in [general] or in the section for your x-lite user. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newb question regarding DTMF
Check the wiki for dtmfmode. It is explained here: http://voip-info.org/tiki-index.php?page=Asterisk%20sip%20dtmfmode -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erik Anderson Sent: Monday, August 23, 2004 7:21 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] newb question regarding DTMF Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newb question regarding DTMF
Hello all - I'm just starting to play around w/ asterisk, and I've run into a seemingly simple problem that has really manged to frustrate me... I'm running the latest cvs version of *, and am trying to dial in to the default extention 1000 demo using x-lite. I can dial and hear the greeting no problem, but when I try and send any DTMF tones, I don't get any response. Is there something specific I need to set in my sip.conf to allow DTMF? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
