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gk at exram.de Guest
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Posted: Wed Apr 29, 2009 12:49 pm Post subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in b |
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First thanks for your reply. I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull. I will try to update to svn trunk tomorrow. Again thanks for first help...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:34) From: Brian West <brian@freeswitch.org> (brian@freeswitch.org) To: gk@exram.de (gk@exram.de) If you subscribe to the event you will receive one on every DTMF press if FreeSWITCH gets it... if you happen to be getting them via inband you won't receive an event unless you enable the inband detection app. brian@freeswitch.org (brian@freeswitch.org) -- Meet us at ClueCon! http://www.cluecon.com |
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brian at freeswitch.org Guest
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Posted: Wed Apr 29, 2009 1:02 pm Post subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in b |
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Well the best option is to NOT use inband at all if possible. And use RFC2833 which eyebeam/xlite support as do most providers out there... You do not HAVE to start_dtmf on sip channels unless they only send the DTMF inband.
set the dtmf-type back to rfc2833 and restart FS.
/b
On Apr 29, 2009, at 12:48 PM, Guido Kuth wrote:
Quote: | First thanks for your reply.
I have subscribed to all Events, so this can't be the mistake. I sent start_dtmf app to FreeSwitch in caller channel and the wiki says that you have to do this on sip channels to enable inband dtmf. I checked sofia.conf and I have found that param dtmf-type is commented out. Would it be helpful to set this to "info"? I think setting it to "rfc2833" would not be very meanigfull.
I will try to update to svn trunk tomorrow.
Again thanks for first help...Guido
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Brian West
brian@freeswitch.org (brian@freeswitch.org)
-- Meet us at ClueCon! http://www.cluecon.com |
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gk at exram.de Guest
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Posted: Wed Apr 29, 2009 1:02 pm Post subject: [Freeswitch-users] Re-2: Serious Problem detecting DTMF in b |
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Hello Anthony, sorry, but I forgot to tell you that I have an inbound ESL connection not an outbound one. So I connect to FS and then wait for Events. I know that I can set async flag in outbound socket, but is this also possible for inbound socket, and when, is it the same as in outbound socket behind the IP-Address? Thank you very much...Guido Original Message processed by David.InfoCenter Subject: Re: [Freeswitch-users] Serious Problem detecting DTMF in bridged SIP call using ESL (29-Apr-2009 19:47) From: Anthony Minessale <anthony.minessale@gmail.com> (anthony.minessale@gmail.com) To: gk@exram.de (gk@exram.de) set the async flag on the socket app call that triggers your ESL connection On Wed, Apr 29, 2009 at 12:21 PM, Guido Kuth <gk@exram.de (gk@exram.de)> wrote: I have a problem I am trying to solve for several days now. I have FS 1.3.0 installed. I have the default configuration except that I have edited event_socket.conf to match my configuration. I have two computers with x-Lite SIP phone 1000 and 1001. Both started and registered. I call in from 1000 and my esl app answers the call plays back a greeting and after that sends a record_session command and a start_dtmf command. Now I send the bridge command with sofia/internal/1001@ip-address ([email]sofia/internal/1001@ip-address[/email]). The x-lite 1001 rings and I can take the call the two can talk to each other and both are able to end the call by hanging up the phone, but there is no reaction on any dtmf tone except when I press * and 1-3, cause this is defined by bind-meta-app in default dialplan. What I need is that I get an Event on DTMF Entry on the bridged call. Please I have to resolve this, cause this is the reason why I came from Asterisk to FreeSwitch. Any help or suggestion is welcome. Thanks in advance...Guido_______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])IRC: sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])pstn:213-799-1400 |
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