Okay i've found sth. I was trying to achieve that by using <action application="bridge" data="{execute_on_answer=start_dtmf}sofia/gateway/OPERATOR_2/$0"/"/> and it failes to convert dtmf While using <action application="bridge" data="{execute_on_answer=start_dtmf}sofia/$0@IP_ADDR_OF_OPERATOR_2"/> and it was success. Is that mean that i my sip profile need to be adjusted somehow? Thx, Mac
Thanks for feedback
Inbound leg has RFC2833, outbound has inband dtmf, so i've changed a part of public.xml as you mentioned: <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <!--<action application="start_dtmf"/>--> <action application="bridge" data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2//$0"/>
Unfortunately it doesn't work.
Thx, Mac
you need to run the start_dtmf() application on the leg that has inband DTMF
if its the inbound leg, add it to the DP <action application="start_dtmf"/>
if it's the outbound leg you need to set it in execute_on_answer in the bridge line
<action application="bridge" data="{execute_on_answer=start_dtmf}sofia/OPERATOR_2/46111223344"/>
On Thu, Feb 24, 2011 at 4:58 PM, Mac <mbsip@gazeta.pl> wrote:
Hi,
Here they are: http://pastebin.freeswitch.org/15473
Thanks, Mac
Get a debug-level log with siptrace enabled and paste it so we can see what's going on.
-Steve
On 22 February 2011 01:16, Mac <mbsip@gazeta.pl> wrote:
Hello,
Could somebody give me a hint?
Thanks, Mac
Hi Steven,
There is much better :) I have different SDPs right now. The one is with rtpmap:101 telephone-event/8000 (leg that is working with RFC2833), the opposite one incorporates a=rtpmap:101 telephone-event/8000 in INVITE SDP and pure codec rtpmaps. There is rtp.p_type == 101 working only on the left side. I cannot find any rtp.p_type == 101 on thark in opposide side which is okay. But the problem still persists.
Once setting my UA to work with Inband DTMF everything works fine.
Thanks, Mac
2011/2/20 Steven Ayre <steveayre@gmail.com>: > <action application="set" data="proxy_media=true"/> > > As indicated by the error, this is the problem. "proxy passthrough". In > proxy mode you pass the media straight through (passthrough). You can't > process passthrough media, such as is needed to mix in inband dtmf. > Use proxy_media=false and bypass_media=false. That's actually the > default so > unless you're setting either to true in the sip profile, you can remove > those lines from the dialplan completely. > Steve on iPhone > On 20 Feb 2011, at 18:51, Mac <mbsip@gazeta.pl> wrote: > > action application="set" data="proxy_media=true"/> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >
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