- FreeSWITCH-users - FreeSWITCH Mailing Lists

Polycom busy lamp with IVR?
by Alex Balashov 16 Jan '15

16 Jan '15

Hi, Sorry if it's a mundane question: I've got two Polycom phones sharing a registration appearance and SLA / busy lamp works fine with them. What I would like is to create a second line appearance whose busy lamp lights up when a call comes into our main IVR. Is this possible? How would I do it? Are there presence hints that can be arbitrarily directed from the dialplan? I wasn't able to discern this from the Polycom presence or general Polycom documentation. Thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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mod_opus bug?
by Thomas Auge 16 Jan '15

16 Jan '15

Hello list, I noticed that in an opus call where the a-leg sends usedtx=0, it is properly forwarded to the b-leg, which in this case also correctly responds with usedtx=0. However, the local SDP for the a-leg then does not have usedtx in the fmtp at all, and it seems to be enabled. My C skills are limited, but could this be a bug: static opus_codec_settings_t default_codec_settings = { .... /*.usedtx */ 1, .... }; By default, usedtx is enabled, correct? However, in static char *gen_fmtp(opus_codec_settings_t *settings, switch_memory_pool_t *pool) there's this: if (settings->usedtx) { snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "usedtx=1; "); } Am I interpreting this correctly that if usedtx is zero, it is omitted in the fmtp line, but then enabled, because that's the default setting? Problem is that Chrome can't handle DTX and generates comfort noise every time it kicks in. Cheers! Thomas P.S.: If my interpretation is correct, it'll also affect useinbandfec.

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User tagging against IP ACL
by Muhammad Naseer Bhatti 16 Jan '15

16 Jan '15

When a call is authenticated via an ACL, is there a way we can flag the user who’s being authenticated? This is needed to billing/identification purpose. Since the user does not exists in the directory and only allowed by an IP ACL, how can we know which user the IP belongs to? — Thanks, Muhammad Naseer Bhatti

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Asterisk's nat=yes feature
by Zoltán Szabó 16 Jan '15

16 Jan '15

Hi, I have a scenario where my softphone is behind nat and freeswitch is not behind nat. With asterisk I could make the audio work with nat=yes in the user's configuration. What is the settings for this on freeswitch? I have tried everything from the following pages: http://wiki.freeswitch.org/wiki/Natted_Softphone_ATA http://wiki.freeswitch.org/wiki/NAT http://wiki.freeswitch.org/wiki/NAT_Traversal http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios (btw these are linking back to each other) Is it not that just easy as net=yes in asterisk? Many thanks, Zoltan

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freeswitch reject calling without registration
by rentmycoder rentmycoder 16 Jan '15

16 Jan '15

Hi all, How is it possible to to reject calling without registration? In asterisk there is no option for this... Thanks, John

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 16 Jan '15

16 Jan '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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Help Needed Debugging Lua Script
by Sina Owolabi 16 Jan '15

16 Jan '15

Hi List, FreeSWITCH newbie here again. I am trying to cobble togther a lua callback script, my first attempt was successful, but I am trying to make it slightly more elegant. I don't see any errors when I try to run this but the callback isnt happening. This is my very second attempt trying to write in lua, so I would be very grateful for any help. The user is expected to dial in, have the call hangup and FreeSWITCH call back. I'm passing a modified $effective_caller_id_number and $destination_number to the lua script: <extension name="callyouback"> <condition field="${caller_id_number}" expression="^1(\d{10})$"require-nested="false"> <action application="set" data="effective_caller_id_number=+234${1}"/> <action application="set" data="effective_caller_id_name=+234${1}"/> </condition> <condition field="destination_number" expression="^012345(6)(7)$"> <action application="set" data="destination_number=+12312345${1}${2}" /> <action application="lua" data="callback.lua ${effective_caller_id_number} ${destination_number}"/> <action application="gentones" data="%(3000,0,430,450,550)"/> <action application="hangup" data="NORMAL_CLEARING"/> </condition> The script itself: api = freeswitch.API(); call_string = "bagpi originate {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" freeswitch.msleep(5000); if (session:ready()) then caller_id_number = session:getVariable("destination_number"); caller_id_name = session:getVariable("destination_number"); number_to_call = session:getVariable("effective_caller_id_number"); api:executeString(call_string); freeswitch.msleep(2000); session:streamFile("/tmp/get_off_my_lawn.wav"); session:hangup("NORMAL_CLEARING"); end

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Video call disconnect after 30 seconds in freeswitch
by Shisheer Teli 16 Jan '15

16 Jan '15

Hi Team, I don't know what happen , but when I start video call it disconnected after every 30 seconds. e.g. x-lite to x-lite call : video call disconnect after 30 seconds X-lite to Zoiper : video call continue, but no video sending. Regards Shisheer T

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working sipp scenario for testing kamailio+FS with authorization
by Борисов, Дмитрий / Dmitriy Borisov 16 Jan '15

16 Jan '15

Have anyone subj? I`m look to RFC and see one, but when i`m looking to sip trace I see another... And it works on SIP-clients, but not on my scenario (in attach) <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="UAC">   <send retrans="500" start_txn="invite"> <![CDATA[ INVITE sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=[field0] 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true" response_txn="invite"/> <recv response="100" optional="true" response_txn="invite"/> <recv response="407" auth="true" response_txn="invite"/> <send ack_txn="invite"> <![CDATA[ ACK sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: [len] v=0 o=[field0] 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <send retrans="500" start_txn="invite"> <![CDATA[ INVITE sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [service] <sip:[field0]@[field1]> Call-ID: [call_id] CSeq: [cseq] INVITE Contact: sip:sipp@[local_ip]:[local_port] [field2] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" optional="true" response_txn="invite"> </recv> <recv response="100" optional="true" response_txn="invite"> </recv> <recv response="183" optional="true" response_txn="invite"/>    <recv response="200" rtd="true" response_txn="invite"> </recv>   <send ack_txn="invite"> <![CDATA[ ACK sip:[field3]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:[field0]@[field1]:[local_port]>;tag=[pid][call_number] To: [field3] <sip:[field3]@[field1]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: [cseq] ACK Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <pause milliseconds="59000"/> <send retrans="500" start_txn="bye"> <![CDATA[ BYE sip:[field0]@[remote_ip]:[remote_port] SIP/2.0 [last_Via:] [last_From:] [last_To:];tag=[pid][call_number] [last_Call-ID:] [last_CSeq:] Contact: <sip:[local_ip]:[local_port];transport=[transport]> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> <recv response="200" crlf="true" response_txn="bye"/>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> </scenario> -- with best regards, Dmitriy Borisov

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Re: [Freeswitch-users] how can i keep A-LEG do not hangup, Vol 103, Issue 199
by ax lyb 16 Jan '15

16 Jan '15

ALL: JUST TRY SOME ideas then i can hold the A-LEG call wait until a free agent to service it, i do this to give a tone_stream after play a wav file, the implement piece of code just as follow: int CallOutProcessBase::processExecuteComplete(const FSBaseMsg &evt, CallTaskObj *pTask,IESLOperation* pCon) { const char* strOriginateUUid = evt.getHeader("variable_origination_uuid"); const char* strApp = evt.getHeader("Application"); const char* strInputKey = evt.getHeader(V_USER_INPUT); //const char* strFlag = evt.getHeader("variable_ax_proc_flag"); if(! strcmp( strApp,"playback")) { // const char* strParam = "1 1 2 2000 1 /usr/local/freeswitch/sounds/input_1.wav /usr/local/freeswitch/sounds/input_1.wav user_input_value ^(1)$" LOG(LS_ERROR)<<"channel playback Complete:"<<strOriginateUUid <<",phone="<<evt.getHeader("Caller-Caller-ID-Number"); //test CallSession* pSession = pTask->find(strOriginateUUid); if( pSession && pSession->state() < E_CST_WAIT_AGENT ) { pCon->execute("playback", "tone_stream://%(2000,4000,440,480);loops=-1", strOriginateUUid,false); pTask->onCallStateChanged(strOriginateUUid, E_CST_WAIT_AGENT , strInputKey); LOG(LS_INFO) << "EXEC PLAYBACK STREAM:"<<pSession->phoneNo() <<","<<pSession->uuids( ); return 0; } LOG(LS_ERROR)<<"we need more agent:"<<pSession->phoneNo()<<","<<pSession->uuids(); return 0; } hope it’s useful to someone same as me! thanks for every one give me instructors > 在 2015年1月16日，10:19，freeswitch-users-request@lists.freeswitch.org 写道： > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio > Quiroz) (ax lyb) > > 发件人: ax lyb <ax.lyb.lei(a)gmail.com> > 收件人: freeswitch-users(a)lists.freeswitch.org > 日期: 2015年1月16日 GMT+810:19:06 > 回复－收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 主题: 回复： [Freeswitch-users] how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) > > > hi, brian,Andrew > > i known a little in lua,i do my customer callcenter in the following style: > > 1. create a inbound connect to see what channel hangup > (BaseFreeswitchClient is a socket-connection wrap) > > int connectFS() > { > //"auth ClueCon\n\n"; > const char* AUTH_ME="ClueCon"; > m_pFsClient = new BaseFreeswitchClient(reinterpret_cast<IChannelDataCallback*>(this)); > if(m_pFsClient->ConnectFS(AUTH_ME) < 0 ) > { > delete m_pFsClient; > m_pFsClient = NULL; > return -1; > } > > // const char* strFilter = "event plain CHANNEL_CREATE \ > // CHANNEL_ANSWER \ > // CHANNEL_BRIDGE \ > // CHANNEL_HANGUP \ > // CHANNEL_HANGUP_COMPLETE\n\n"; > const char* strFilter = "event plain CHANNEL_HANGUP_COMPLETE\n\n"; > > if( m_pFsClient->filterEvent(strFilter) < 0 ) > { > delete m_pFsClient ; > m_pFsClient = NULL; > return -1; > } > return 0; > } > > 2. create a server for outbound connect, the outbound connect for every call-session to FILTER > CHANNEL_ANSWER,CHANNEL_HANGUP_COMPLETE,CHANNEL_EXECUTE_COMPLETE > > case ESL_EVENT_CHANNEL_DATA: > { > const char* strEvtName = evt.getHeader("Event-Name"); > > char szFilter[1024]; > const char* strFilter = "event plain CHANNEL_ANSWER CHANNEL_BRIDGE CHANNEL_EXECUTE_COMPLETE DTMF\n\n"; > snprintf(szFilter, sizeof(szFilter),"%sfilter Unique-ID %s\n\n",strFilter, > evt.getHeader("Caller-Unique-ID")); > int nWriteSize = strnlen(szFilter,sizeof(szFilter)); > LOG(LS_ERROR)<<strEvtName << "," <<evt.getHeader( "Caller-Unique-ID") > <<",nWriteSize="<<nWriteSize <<","<<szFilter; > p->onWrite(szFilter, nWriteSize); > } > > 3. set "session.SetAutoHangup(false);”, which channel_variable should i set? > > > thanks > > >> 在 2015年1月16日，08:18，freeswitch-users-request@lists.freeswitch.org <mailto:freeswitch-users-request@lists.freeswitch.org> 写道： >> >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request(a)lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner(a)lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> Today's Topics: >> >> 1. Re: Help Needed Debugging Lua Script (Luis Daniel Lucio Quiroz) >> 2. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) >> >> 发件人: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> >> 收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> 日期: 2015年1月16日 GMT+808:13:15 >> 回复－收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> 主题: 回复： [Freeswitch-users] Help Needed Debugging Lua Script >> >> >> Don't loose your time, contact me offline >> >> On Jan 15, 2015 5:49 PM, "Sina Owolabi" <notify.sina(a)gmail.com <mailto:notify.sina@gmail.com>> wrote: >> Hi List, >> >> I think I have finally come up with something that works, the script captures the details correctly, but I wonder if there is a better way to write it? I am trying to get it to call the dialer back and play a message, which it does correctly. >> Please what can I do to make it to retry the dialler's number in case the call doesn't go through the first (say) two times, just in case the dialler cut it off mistakenly, or the telco plays a repeated message after freeswitch hung up, preventing the callback from coming through? (I've seen this happen a few times with a local telco). Thanks! >> >> number_to_call = argv[1]; >> caller_id = argv[2]; >> api = freeswitch.API(); >> dialString = "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; >> session1 = freeswitch.Session(dialString); >> session1:sleep(5000); >> >> if (session1:ready()) then >> session1:sleep(35000); >> api:execute("bgapi originate", "session1"); >> session1:sleep(3000); >> session1:streamFile('/tmp/stop_calling_me_stalker.wav'); >> session1:hangup("NORMAL_CLEARING"); >> end >> >> >> On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus <avi(a)avimarcus.net <mailto:avi@avimarcus.net>> wrote: >> Two things: >> 1) You aren't grabbing the arg, but the channel variable.. try this in your script: >> caller_id_number = argv[1] >> number_to_call = argv[2] >> >> 2) I don't think you're managing your hangup/callback originate properly. >> I don't think you want to use bgapi... or maybe you just need a destination. It's "originate <http://wiki.freeswitch.org/wiki/Mod_commands#originate> sofia/A endpoint" -- you need to specify where it goes to, the lua script can't "receive" the call. You can have it received by e.g: &lua(pickup.lua) >> >> api = freeswitch.API() >> api:execute("originate", DialString.." &lua(pickup.lua)"); >> >> Also: >> Maybe you want to use it as a hangup hook. Instead of: >> <action application="lua" data="callback.lua ${effective_caller_id_number} ${destination_number}"/> >> Do: >> <action application="set" data="api_hangup_hook=lua callback.lua ${effective_caller_id_number} ${destination_number}"/> >> <action application="hangup" data="486"/> >> >> -Avi >> >> On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi <notify.sina(a)gmail.com <mailto:notify.sina@gmail.com>> wrote: >> Hi List, >> >> FreeSWITCH newbie here again. >> I am trying to cobble togther a lua callback script, my first attempt >> was successful, but I am trying to make it slightly more elegant. >> I don't see any errors when I try to run this but the callback isnt happening. >> This is my very second attempt trying to write in lua, so I would be >> very grateful for any help. >> >> The user is expected to dial in, have the call hangup and FreeSWITCH call back. >> >> I'm passing a modified $effective_caller_id_number and >> $destination_number to the lua script: >> >> <extension name="callyouback"> >> <condition field="${caller_id_number}" >> expression="^1(\d{10})$"require-nested="false"> >> <action application="set" data="effective_caller_id_number=+234${1}"/> >> <action application="set" data="effective_caller_id_name=+234${1}"/> >> </condition> >> <condition field="destination_number" expression="^012345(6)(7)$"> >> <action application="set" data="destination_number=+12312345${1}${2}" /> >> <action application="lua" data="callback.lua >> ${effective_caller_id_number} ${destination_number}"/> >> <action application="gentones" data="%(3000,0,430,450,550)"/> >> <action application="hangup" data="NORMAL_CLEARING"/> >> </condition> >> >> >> The script itself: >> >> api = freeswitch.API(); >> call_string = "bagpi originate >> {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" >> >> freeswitch.msleep(5000); >> if (session:ready()) then >> caller_id_number = session:getVariable("destination_number"); >> caller_id_name = session:getVariable("destination_number"); >> number_to_call = session:getVariable("effective_caller_id_number"); >> >> api:executeString(call_string); >> freeswitch.msleep(2000); >> session:streamFile("/tmp/get_off_my_lawn.wav"); >> session:hangup("NORMAL_CLEARING"); >> end >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> >> >> >> 发件人: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> >> 收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> 日期: 2015年1月16日 GMT+808:18:10 >> 回复－收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >> 主题: 回复： [Freeswitch-users] how can i keep A-LEG do not hangup >> >> >> Read about queues >> >> On Jan 15, 2015 11:53 AM, "Brian West" <brian(a)freeswitch.org <mailto:brian@freeswitch.org>> wrote: >> Thats not relevant in this case, The autoHangup is when used in say lua... >> >> This case you will probably want this combo: >> >> https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_e… <https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_e…> >> https://wiki.freeswitch.org/wiki/Variable_park_after_bridge <https://wiki.freeswitch.org/wiki/Variable_park_after_bridge> >> >> >> >> >> On Thu, Jan 15, 2015 at 7:27 AM, Andrew <aademattia(a)comcast.net <mailto:aademattia@comcast.net>> wrote: >> This may be able to help you but I set session.SetAutoHangup(false); >> >> This will stop the dial plan from ending. >> >> <> >> From: freeswitch-users-bounces(a)lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of ax lyb >> Sent: Thursday, January 15, 2015 2:52 AM >> To: freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> >> Subject: [Freeswitch-users] how can i keep A-LEG do not hangup >> >> >> >> All: >> >> recently i write a small program based freeswitch, in this program i only do things like callcenter mod, >> >> here is what i do: >> >> >> >> 1. originate a call (“ originate sofia/gateway/gw01/xxxx my-out-call XML default”), in the default.xml, >> >> i already config it as follow: >> >> >> >> <extension name=“my-call-out"> >> >> 65 <condition field="destination_number" expression="^my-call-out$"> >> >> 66 <action application="socket" data="127.0.0.1:9600 <http://127.0.0.1:9600/> async full"/> >> >> 67  >> >> 72 </condition> >> >> 73 </extension> >> >> >> >> 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , >> >> bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, >> >> these steps run normally ok. >> >> >> >> 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, >> >> i’ll leave the other call do nothing,for i can not get more free agents to service; some special scene >> >> is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only >> >> A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it’s not user hangup manually), >> >> how can i do make the C3 call not hangup until i get a free agent to service it? >> >> >> >> >> >> any suggest is appreciate, >> >> >> >> >> >> >> >> ax.lyb.lei(a)gmail.com <mailto:ax.lyb.lei@gmail.com> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> >> >> >> -- >> Brian West >> brian(a)freeswitch.org <mailto:brian@freeswitch.org> >> >> Twitter: @FreeSWITCH , @briankwest >> http://www.freeswitchbook.com <http://www.freeswitchbook.com/> >> http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> >> T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) >> iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> >> http://www.freeswitch.org <http://www.freeswitch.org/> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > ax lyb > ax.lyb.lei(a)gmail.com <mailto:ax.lyb.lei@gmail.com> > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ax lyb ax.lyb.lei(a)gmail.com

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