[Freeswitch-users] Bypass Media not work

[Anthony Minessale]

did you set it first in your dialplan before anything else and make
sure your dialplan does nothing at all that reqires media.
Ideally nothing but setting the var and the bridge line.

<action application="set" data="bypass_media=true"/>
<action application="bridge" data="user/1004"/>

if you do anything at all that requires media from FS it will about
the bypass and establish media from FS.

It works, once you tell yourself that and look for how you have
misconfigured it, then you will have success.



On Tue, Apr 24, 2012 at 10:55 AM, Cucu Florian <koralu at gmail.com> wrote:
> Thank you for the replies but didn't work.
>
> I uncomment late-negotioation parameter in sip_profile/internal.xml
> <param name="inbound-late-negotiation" value="true"/>
>
> I put <action application="set" data="bypass_media=true"/> in my dial plan.
>
> I set for both Xlite only iLBC codec. The logs are:
>
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [iLBC:98:8000:30:13330]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4827 Audio Codec
> Compare [telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
> 2012-04-24 18:48:58.207474 [DEBUG] sofia_glue.c:4948 Set 2833 dtmf
> send/recv payload to 101
> 2012-04-24 18:48:58.207474 [DEBUG] switch_channel.c:2848
> (sofia/internal/1001 at floflo85.zapto.org) Callstate Change DOWN ->
> HANGUP
> 2012-04-24 18:48:58.207474 [DEBUG] switch_core_state_machine.c:362
> (sofia/internal/1001 at floflo85.zapto.org) Running State Change
> CS_HANGUP
> 2012-04-24 18:48:58.207474 [NOTICE] sofia.c:5805 Hangup
> sofia/internal/1001 at floflo85.zapto.org [CS_NEW]
> [INCOMPATIBLE_DESTINATION]
>
>  When I set for just one Xlite iLBC and also G711 the call is ok.
>
> Ty.
>
> On 4/24/12, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> To use bypass from the bypassl_media=true variable you must enable
>> inbound-late-negotiation in your sofia profile.
>>
>>
>> On Tue, Apr 24, 2012 at 8:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>>> You still need to have the codecs enabled in your FreeSWITCH
>>> configuration,
>>> also FreeSWITCH supports iLBC and Broadvoice32 and 16 codecs along with a
>>> list of other codecs that are just not enabled by default...
>>>
>>> K
>>>
>>>
>>>
>>>
>>> On 4/24/12 5:19 AM, "Cucu Florian" <koralu at gmail.com> wrote:
>>>
>>> Hello,
>>>
>>> I tried to test the behavior of "bypass media" in FS with the following
>>> scenario: Xlite(ext.1000) -> FS <- Xlite(ext:1001)
>>>
>>> On each Xlite I set one single codec (that is currently unsupported by FS)
>>> as it said in wiki page: "This is useful if you have two end-points that
>>> need to use a codec that is currently not supported in FreeSWITCH "
>>>
>>> As the FS tutorial said "Before executing the bridge action you must set
>>> the
>>> "bypass_media" flag to true. " so I put the following line in my dial
>>> plan.
>>>
>>> <extension name="extension-intercom">
>>>     <condition field="destination_number" expression="1001">
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge" data="user/1001"/>
>>>     </condition>
>>> </extension>
>>>
>>>
>>> <extension name="extension-intercom">
>>>     <condition field="destination_number" expression="1000">
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge" data="user/1000"/>
>>>     </condition>
>>> </extension>
>>>
>>> When I put in Xlite also G711ulaw/alaw as codecs preference the calls was
>>> ok. But when I put only codecs that FS do not support(ex.
>>> iLBC,BroadVoice32)
>>> the call failed. What am I missing?
>>>
>>> I made this scenario because I need to route via FS some calls with G729r8
>>> codec and I need no codec negotiation.
>>>
>>> Ty
>>>
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>>>
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>>>
>>> 
>>> 
>>>
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>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
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>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
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>> http://www.freeswitch.org
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900