Re: [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.

2010-03-12 Thread Kevin P. Fleming
Joakim Eriksson wrote:
 I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
 When a user calls from skype (not skype-in) to asterisk, dtmf (basically 
 menus for a conference system) works just fine.
 But when a user from the inside (soft or hardware sip phone) calls out via 
 skype-out dtmf doesn't work.
 I have tried setting the codec to alaw, and dtmfmode to all possible options 
 (auto, inband and rfc2833).

This is a known issue with SkypeIn and SkypeOut and is being addressed.
There should be a Skype For Asterisk release soon that contains the
changes required on its send; there are also changes being made in the
SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-)

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Asterisk 1.6.2.5 x64 with Skype and DTMF on skype-out.

2010-03-12 Thread Joakim Eriksson
Thank for the help :)
Then i can just hope it gets fixed soon.
(But now that i know about it, its not as critical anymore). 

//Joakim

On Mar 12, 2010, at 8:24 PM, Kevin P. Fleming wrote:

 Joakim Eriksson wrote:
 I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform.
 When a user calls from skype (not skype-in) to asterisk, dtmf (basically 
 menus for a conference system) works just fine.
 But when a user from the inside (soft or hardware sip phone) calls out via 
 skype-out dtmf doesn't work.
 I have tried setting the codec to alaw, and dtmfmode to all possible options 
 (auto, inband and rfc2833).
 
 This is a known issue with SkypeIn and SkypeOut and is being addressed.
 There should be a Skype For Asterisk release soon that contains the
 changes required on its send; there are also changes being made in the
 SkypeIn and SkypeOut networks to properly support DTMF. Stay tuned :-)
 
 -- 
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org
 
 -- 
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users