One way DTMF Gizmo/SIPPhone <-> Asterisk 1.4

[jsgoecke]

I am getting one-way DTMF working with Gizmo/SIPPhone and Asterisk v1.4. The calls connect, audio is good both ways, but when I dtmf from the phone on the Asterisk side, nothing comes across to Gizmo. When I DTMF from the Gizmo client, I get it on the Asterisk.

I have my [general] setting for sip.conf set to ‘dtmfmode=rfc2833’, so it should be working as per the knowledgebase. Any ideas?

Without this, the new Grandcentral feature of going via Gizmo to my softphone and Asterisk won’t fly, as of course I can not answer a Grandcentral call without DTMF.

[AndrewZ]

As far as I remember it was suggested somethere that dtmfmode should be specified per trunk as well.
At the same time it will be good to see some debug output to check if other side is really advertizing it’s rfc2833 support. You may also try to set dtmfmode=info for that trunk.

[jsgoecke]

Turned out it was a problem on the end point (Linksys SPA942), sending inband instead of RFC2833. Strange that Asterisk does not do this translation…thought it would.

[valley]

Might not be a good idea for Asterisk to listen to audio by default, especially with all this canreinvite business.

[jsgoecke]

@valley

Might not be a good idea for Asterisk to listen to audio by default, especially with all this canreinvite business.

I don’t allow canreinvite on this instance for a myriad of reasons. But this was not the issue on this particular issue.

[valley]

@muppetmaster

Turned out it was a problem on the end point (Linksys SPA942), sending inband instead of RFC2833. Strange that Asterisk does not do this translation…thought it would.

Or it does. Just bumpted into this while reading

@http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not available in Asterisk 1.0.x

[jsgoecke]

@valley

[quote=“muppetmaster”]Turned out it was a problem on the end point (Linksys SPA942), sending inband instead of RFC2833. Strange that Asterisk does not do this translation…thought it would.

Or it does. Just bumpted into this while reading

@http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode

auto: Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. This feature was added on Sep 6, 2005 and is not available in Asterisk 1.0.x[/quote]

I don’t follow how negotiated endpoint support of codecs/dtmf is related to canreinvite in terms of the issue originally stated in this post. Especially when it was established that indeed the Linksy SPA942 I am using was not using RFC2833 until I made it the only option in its config.