Hi, Which DTMF mode do people mostly use? I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones (for feature usage), the tones arent repeated to the end user. So if I call a company that has a menu system, I can't use the menu. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101013/939d217b/attachment.htm
It depends upon whether you are receiving DTMF or sending, and whether you
are using a VoIP protocol or using DAHDI/Zaptel.
Could you explain a bit what type of setup you have?
Zeeshan A Zakaria
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www.ilovetovoip.com
On 2010-10-13 9:15 AM, "Dan Journo" <dan at
keshercommunications.com> wrote:
Hi,
Which DTMF mode do people mostly use?
I've tried SIP INFO and RFC2833 but although Asterisk recognises the tones
(for feature usage), the tones arent repeated to the end user.
So if I call a company that has a menu system, I can't use the menu.
Thanks
Dan
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* The provider has confirmed that they support rfc2833 or inband with the right codecs. Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101013/a3264bd2/attachment.htm
> Are you using the .gsm codec or some other flavor (ulaw, alaw, G729?)This is from the sip.conf for the provider: allow=gsm allow=ulaw This is from the sip extension:- alaw,ulaw,gsm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101013/848db1ba/attachment.htm
I would suggest first to make sure that asterisk is receiving DTMF fine from
your IP devices/phones. Do you have a test IVR where you can dial and press
digits and verify that asterisk is responding?
Once you are sure that asterisk is receiving DTMF fine, then you should ask
your provider what DTMF setting you should have on your system. Usually all
of them support RFC2833, so if in your sip.conf where you have defined the
trunk, dtmfmode is set to rfc2833, your provider should receive it and pass
on to the next carrier or trunk.
Zeeshan A Zakaria
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www.ilovetovoip.com
On 2010-10-13 10:19 AM, "Dan Journo" <dan at
keshercommunications.com> wrote:
> It depends upon whether you are receiving DTMF or sending, and whether
you are using a VoIP protoc...
Sorry about the lack of info.
It's a simple SIP only setup. A handful of sip phones, an asterisk server,
and a sip provider.
The DTMF signals from the sip phones are received by Asterisk because they
can access features like *1.
The DTMF signal from the called party are received by Asterisk because they
can also access features like *1.
But, the DTMF tones are not passed through from the Sip Phone to the Called
Party.
The same happens regardless of whether its an incoming or outgoing call.
That means, if any of my users try to call a company with a menu system,
they can't select any options.
How can I tell if Asterisk is sending the tones through to the provider? I
need to find out whether its something I'm doing, or something the provider
is doing.
Any ideas?
Thanks
Dan
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> I would suggest first to make sure that asterisk is receiving DTMF fine from your IP devices/phones. Do you have a test IVR where you can dial and press digits and verify that asterisk is responding?Made a quick IVR, and its working for both sides of the asterisk (between the provider and asterisk, and between the sip phones and asterisk). I think its an issue with DTMF Pass-through. Is there a way to disable DMTF passthrough? Maybe asterisk is blocking the signals from being repeated to the other party? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101013/9693dd5d/attachment.htm
Could features.conf be preventing asterisk from repeating the DTMF tones?
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode Could features.conf be preventing asterisk from repeating the DTMF tones? Perhaps. What is your featuredigittimeout value?
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode> Since alaw is "sort of making it work", either your SIP provider or yourother party is not in U.S. I dont understand why the codec should make a difference if im using rfc2833. Could you clear that up for me?
> From what I read, the codec could be trying to switch from rfc2833 to inbandduring the call, causing the "stuck" effect. Any way to prevent that?
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Journo Sent: Wednesday, October 13, 2010 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DMTF Mode> From what I read, the codec could be trying to switch from rfc2833 toinband during the call, causing the "stuck" effect. Any way to prevent that? According to the WIKI, changing rfc2833 to auto in sip.conf should do the trick.
> Since alaw is "sort of making it work", either your SIP provider or yourother party is not in U.S. Its stopped working again. This is really unusual. I didnt change anything. I decided to do a tcpdump, and I can clearly see the rfc2833 packets being exchanged correctly. Why should both parties not be able to hear the tones?