Re: [Freeswitch-users] [ringback] problems with dingaling
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
On Jan 6, 2009, at 4:30 AM, kriko wrote:
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.
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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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:
On Jan 6, 2009, at 4:30 AM, kriko wrote:
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
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
you may want to try <action application="bridge" data="{ignore_early_media=true}dingaling/ gmail.com/$1@gmail.com"/> jingle has no concept of telephony early media waiting for answer and all that so it's not an exact fit into sip. On Thu, Jan 8, 2009 at 7:32 AM, kriko <kristjan.ugrin@gmail.com> wrote:
Now I've made a small dialplan to call from sip phone directly to gtalk:
<extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$"> <!-- <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/$1@gmail.com "/> </condition> </extension>
Simple, calling works. However still can't get ringback to work. In this case the first leg is not yet aswered. If I apply same stuff onto SIP to SIP call then ringback works. Dingaling problem?
Log: http://pastebin.com/m37354677
This is all that 2009-01-08 14:25:56 [NOTICE] mod_dingaling.c:1110 negotiate_media() Ring-Ready dingaling/gmail.com/atomic.devterium@gmail.com! 2009-01-08 14:25:56 [DEBUG] mod_dingaling.c:1058 do_describe() Don't have my codec yet here's one
-- kriko
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Nope. Currently only gtalk → sip ringback works, sip → gtalk doesn't. If soemone needs, I'm pasting my extensions used. sip → gtalk (ringback not working): <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$"> <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:
you may want to try
<action application="bridge" data="{ignore_early_media=true}dingaling/ gmail.com/$1@gmail.com"/>
jingle has no concept of telephony early media waiting for answer and all that so it's not an exact fit into sip.
On Thu, Jan 8, 2009 at 7:32 AM, kriko <kristjan.ugrin@gmail.com> wrote:
Now I've made a small dialplan to call from sip phone directly to gtalk:
<extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^gmail\+([^\@]+)\@?(.*)$"> <!-- <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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-- kriko
participants (3)
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Anthony Minessale
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Brian West
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kriko