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[Freeswitch-users] [ringback] problems with dingaling


 
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kristjan.ugrin at gmai...
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PostPosted: Tue Jan 06, 2009 5:31 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

I'm trying to entertain caller with some music or audio message so he knows that call is in progress.
I tried to setup ringback like this:
"api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221 &bridge(sofia/default/1000@10.99.8.221)"

and it works. I can hear continuous playback of this wav file, however when calling a gtalk user:
"api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221 &bridge(dingaling/gmail.com/my.mail@gmail.com)"

there is silence only. Is this dingaling problem? Could it be solved?

--
kriko



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brian at freeswitch.org
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PostPosted: Tue Jan 06, 2009 9:52 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

On Jan 6, 2009, at 4:30 AM, kriko wrote:

Quote:
I'm trying to entertain caller with some music or audio message so
he knows that call is in progress.
I tried to setup ringback like this:
"api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221
&bridge(sofia/default/1000@10.99.8.221)"'

In this case the first leg is already answered you need to use
transfer_ringback.


Quote:


and it works. I can hear continuous playback of this wav file,
however when calling a gtalk user:
"api originate {ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221
&bridge(dingaling/gmail.com/my.mail@gmail.com)"

there is silence only. Is this dingaling problem? Could it be solved?

--
kriko



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kristjan.ugrin at gmai...
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PostPosted: Wed Jan 07, 2009 8:59 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

originate {transfer_ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221 &bridge(sofia/default/1000@10.99.8.221)
This works, but still when calling gtalk user:
originate {transfer_ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}user/1003@10.99.8.221 &bridge(dingaling/gmail.com/atomic.devterium@gmail.com)

there is only dead silence. I would be glad even with the default ringing tone.

Log:
http://pastebin.com/m7a0d6a9e

Maybe it is because of dingaling? When originating a call between softphones, ringing tone or ringback is working.

On Tue, 06 Jan 2009 15:48:52 +0100, Brian West <brian@freeswitch.org> wrote:

Quote:

On Jan 6, 2009, at 4:30 AM, kriko wrote:

Quote:
I'm trying to entertain caller with some music or audio message so
he knows that call is in progress.
I tried to setup ringback like this:
"api originate
{ringback=/usr/sounds/en/us/callie/misc/16000/we_are_trying_to_reach.wav}sofia/default/1003@10.99.8.221
&bridge(sofia/default/1000@10.99.8.221)"'

In this case the first leg is already answered you need to use
transfer_ringback.





--
kriko



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kristjan.ugrin at gmai...
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PostPosted: Thu Jan 08, 2009 8:38 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

Now I've made a small dialplan to call from sip phone directly to gtalk:

<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$">
<!-- <action application="answer"/> -->
<!-- <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/>
</condition>
</extension>

Simple, calling works. However still can't get ringback to work. In this case the first leg is not yet aswered.
If I apply same stuff onto SIP to SIP call then ringback works. Dingaling problem?

Log:
http://pastebin.com/m37354677

This is all that
2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() Ring-Ready dingaling/gmail.com/atomic.devterium@gmail.com!
2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't have my codec yet here's one




--
kriko



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anthony.minessale at g...
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PostPosted: Thu Jan 08, 2009 9:12 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

you may want to try

<action application="bridge" data="{ignore_early_media=true}dingaling/gmail.com/$1@gmail.com"/>

jingle has no concept of telephony early media waiting for answer and all that so it's not an exact fit into sip.


On Thu, Jan 8, 2009 at 7:32 AM, kriko <kristjan.ugrin@gmail.com (kristjan.ugrin@gmail.com)> wrote:
Quote:
Now I've made a small dialplan to call from sip phone directly to gtalk:

<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$">
<!-- <action application="answer"/> -->
<!-- <action application="playback" data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/>
</condition>
</extension>

Simple, calling works. However still can't get ringback to work. In this case the first leg is not yet aswered.
If I apply same stuff onto SIP to SIP call then ringback works. Dingaling problem?

Log:
http://pastebin.com/m37354677

This is all that
2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() Ring-Ready dingaling/gmail.com/atomic.devterium@gmail.com!
2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't have my codec yet here's one





--
kriko



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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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kristjan.ugrin at gmai...
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PostPosted: Thu Jan 08, 2009 9:21 am    Post subject: [Freeswitch-users] [ringback] problems with dingaling Reply with quote

Nope.
Currently only gtalk → sip ringback works, sip → gtalk doesn't.

If soemone needs, I'm pasting my extensions used.


sip → gtalk (ringback not working):
<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$">
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/>
<action application="set" data="hangup_after_bridge=true"/>
<!-- <action application="bridge" data="dingaling/gmail.com/$1@gmail.com"/> -->
<action application="bridge" data="{ignore_early_media=true}dingaling/gmail.com/$1@gmail.com"/>
</condition>
</extension>

gtalk → sip:
<extension name="gmail2sip">
<condition field="caller_id_number" expression="^([^@]+)" break="never">
<!--Nokia bug - problems with @ in caller_id -->
<action application="set" data="effective_caller_id_number=$1"/>
<action application="set" data="effective_caller_id_name=$1@gmail.com"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="ringback=tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/$${domain}/${destination_number}"/>
</condition>
</extension>


Thanks for your help!

On Thu, 08 Jan 2009 15:09:41 +0100, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Quote:
you may want to try

<action application="bridge" data="{ignore_early_media=true}dingaling/
gmail.com/$1@gmail.com"/>

jingle has no concept of telephony early media waiting for answer and all
that so it's not an exact fit into sip.


On Thu, Jan 8, 2009 at 7:32 AM, kriko <kristjan.ugrin@gmail.com> wrote:

Quote:
Now I've made a small dialplan to call from sip phone directly to gtalk:

<extension name="sip2jingle">
<condition field="source" expression="mod_sofia"/>
<condition field="destination_number"
expression="^gmail\+([^\@]+)\@?(.*)$">
<!-- <action application="answer"/> -->
<!-- <action application="playback"
data="tone_stream://path=${base_dir}/conf/tetris.ttml;loops=10"/> -->
<action application="set" data="continue_on_fail=true"/>
<action application="set"
data="ringback=%(2000,4000,440.0,480.0)"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="dingaling/gmail.com/$1@gmail.com
"/>
</condition>
</extension>

Simple, calling works. However still can't get ringback to work. In this
case the first leg is not yet aswered.
If I apply same stuff onto SIP to SIP call then ringback works.
Dingaling
problem?

Log:
http://pastebin.com/m37354677

This is all that
2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media()
Ring-Ready dingaling/gmail.com/atomic.devterium@gmail.com!
2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't
have
my codec yet here's one




--
kriko



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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org







--
kriko



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