i was trying to ring multiple endpoints (A calls B1 and B2 simultaneously) by doing <action application="bridge" data="user/${callee1},user/${callee2}"/> i was expecting one callee hung up once the other answers the call, but what i was experiencing is one hung up once the other rings, the FS console log says: [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1@1.2.3.4! [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2 [CS_CONSUME_MEDIA] [LOSE_RACE] so how can i make the calls happen as expected?
Hi Vincent, the action looks correct, what else is in this extension? Also, (this is a trap that one of my users created) are there any active forward or divert settings on the endpoints themselves? - Francis On 23/05/2014 8:34 PM, Vincent Xia wrote:
i was trying to ring multiple endpoints (A calls B1 and B2 simultaneously) by doing <action application="bridge" data="user/${callee1},user/${callee2}"/> i was expecting one callee hung up once the other answers the call, but what i was experiencing is one hung up once the other rings, the FS console log says:
[NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1@1.2.3.4! [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2 [CS_CONSUME_MEDIA] [LOSE_RACE]
so how can i make the calls happen as expected?
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the entire dialplan extension is something like: <extension name="simul_call" continue="false"> <condition field="destination_number" expression="^(\b1\d{3}\b)$"> <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="transfer_ringback=$${hold_music}" /> <action inline="true" application="lua" data="get_callee.lua" /> </condition> <condition field="${is_call_allowed}" expression="allowed" continue="false"> <action application="bridge" data="${callee1},${callee2}" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> (the variables callee1 and callee2 are assigned within get_callee.lua) it works fine with multiple linphone endpoints, but not with some others as sipdroid, maybe in these scenarios the FS is acting differently against 180 and 183 response from the endpoints, any ideas? 2014-05-24 19:56 GMT+08:00 Francis <sms@icefire.qza.net.au>:
Hi Vincent, the action looks correct, what else is in this extension? Also, (this is a trap that one of my users created) are there any active forward or divert settings on the endpoints themselves?
- Francis
On 23/05/2014 8:34 PM, Vincent Xia wrote:
i was trying to ring multiple endpoints (A calls B1 and B2 simultaneously) by doing <action application="bridge" data="user/${callee1},user/${callee2}"/> i was expecting one callee hung up once the other answers the call, but what i was experiencing is one hung up once the other rings, the FS console log says:
[NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1@1.2.3.4! [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2 [CS_CONSUME_MEDIA] [LOSE_RACE]
so how can i make the calls happen as expected?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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Try adding this: <action application="set" data="ignore_early_media=true"/> Looking at the log snippet you posted earlier, I suspect this *might* be the cause. On 2014-05-26 12:57, Vincent Xia wrote:
the entire dialplan extension is something like: <extension name="simul_call" continue="false"> <condition field="destination_number" expression="^(b1d{3}b)$"> <action application="bind_meta_app" data="1 ab s execute_extension::dx XML features" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="transfer_ringback=$${hold_music}" /> <action inline="true" application="lua" data="get_callee.lua" /> </condition> <condition field="${is_call_allowed}" expression="allowed" continue="false"> <action application="bridge" data="${callee1},${callee2}" /> <action application="answer" /> <action application="sleep" data="1000" /> </condition> </extension> (the variables callee1 and callee2 are assigned within get_callee.lua)
it works fine with multiple linphone endpoints, but not with some others as sipdroid, maybe in these scenarios the FS is acting differently against 180 and 183 response from the endpoints, any ideas?
2014-05-24 19:56 GMT+08:00 Francis <sms@icefire.qza.net.au>:
Hi Vincent, the action looks correct, what else is in this extension? Also, (this is a trap that one of my users created) are there any active forward or divert settings on the endpoints themselves?
- Francis
On 23/05/2014 8:34 PM, Vincent Xia wrote:
i was trying to ring multiple endpoints (A calls B1 and B2 simultaneously) by doing <action application="bridge" data="user/${callee1},user/${callee2}"/> i was expecting one callee hung up once the other answers the call, but what i was experiencing is one hung up once the other rings, the FS console log says:
[NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/sip:callee1@1.2.3.4! [NOTICE] switch_ivr_originate.c:3448 Hangup sofia/callee2 [CS_CONSUME_MEDIA] [LOSE_RACE]
so how can i make the calls happen as expected?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com [1]
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [2]
Official FreeSWITCH Sites http://www.freeswitch.org [3] http://wiki.freeswitch.org [4] http://www.cluecon.com [5]
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [6] UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users [7] http://www.freeswitch.org [3]
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com [1]
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [2]
Official FreeSWITCH Sites http://www.freeswitch.org [3] http://wiki.freeswitch.org [4] http://www.cluecon.com [5]
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com [1]
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com [2]
Official FreeSWITCH Sites http://www.freeswitch.org [3] http://wiki.freeswitch.org [4] http://www.cluecon.com [5]
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Links: ------ [1] http://www.freeswitchsolutions.com [2] http://www.cudatel.com [3] http://www.freeswitch.org [4] http://wiki.freeswitch.org [5] http://www.cluecon.com [6] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [7] http://lists.freeswitch.org/mailman/options/freeswitch-users
Hi everyone ! I'm not familiar with freeswitch, I'm a web developer, and my company uses this solution for VoIP calls. I read the cookbook to understand how it works. I have to develop a solution where we can see who calls who in a live interface, but I'm a little bit lost about how to get this information from PHP ESL. Any help to show me the way to go would be appreciated. Thank you. Emrah K.
you can watch events to pull this information live via php. Mike On May 26, 2014, at 7:30 AM, Emrah <emrah.karademir@tellpemail.com> wrote:
Hi everyone !
I’m not familiar with freeswitch, I’m a web developer, and my company uses this solution for VoIP calls.
I read the cookbook to understand how it works.
I have to develop a solution where we can see who calls who in a live interface, but I’m a little bit lost about how to get this information from PHP ESL.
Any help to show me the way to go would be appreciated.
Thank you. Emrah K.
Thank you for your answer, the penny just dropped and now I know what to do. Best regards, Emrah K. _____ De : freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] De la part de Michael Jerris Envoyé : mardi 27 mai 2014 15:47 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] PHP ESL you can watch events to pull this information live via php. Mike On May 26, 2014, at 7:30 AM, Emrah <emrah.karademir@tellpemail.com> wrote: Hi everyone ! Im not familiar with freeswitch, Im a web developer, and my company uses this solution for VoIP calls. I read the cookbook to understand how it works. I have to develop a solution where we can see who calls who in a live interface, but Im a little bit lost about how to get this information from PHP ESL. Any help to show me the way to go would be appreciated. Thank you. Emrah K.
participants (4)
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Emrah
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Francis
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Michael Jerris
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Vincent Xia