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2 CS_DESTROY events without trying the second one in the bridge
by Muhammad Naseer Bhatti 27 Nov '16

27 Nov '16

I have a simple Dialplan with two gateways to try, <condition field='destination_number' expression='$'> <action application='set' data='hangup_after_bridge=true'/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="disable_hold=true"/> <action application="set" data="map_type=pbx"/> <action application="set" data="map_id=10"/> <action application="set" data="state=NC"/> <action application="set" data="npa=910"/> <action application="set" data="export_vars=map_type,map_id,state,npa,dispatcher_ip"/> <action application="set" data="import=dispatcher_ip"/> <action application="set" data="process_cdr=b_only"/> <action application="set" data="bypass_media=true"/> <action application="limit_execute" data="hash default 5.9.50.206:5060 3 bridge [dispatcher_ip=5.9.50.206]sofia/external/92301@5.9.50.206:5060 |[dispatcher_ip=10.211.55.26]sofia/external/92301@10.211.55.26:15080"/> </condition> and I have tried to bridge the same with 2 lines of execute with hangup_after_bridge set to true. <action application="limit_execute" data="hiredis default 5.9.50.206:5060 3 bridge [dispatcher_ip=5.9.50.206]sofia/external/ 92301@5.9.50.206:5060"/> <action application="limit_execute" data="hiredis default 10.211.55.26:15080 1 bridge [dispatcher_ip=10.211.55.26]sofia/external/92301@10.211.55.26:15080”/> For testing, the first IP in the bridge is not reachable, so I cancel the call after a few seconds. 2016-11-26 12:56:37.368968 [NOTICE] sofia.c:8234 Hangup sofia/external/ 91000(a)10.211.55.12 [CS_EXECUTE] [ORIGINATOR_CANCEL] which is fine, but the CS_DESTROY event is generated twice. Pastebin @ https://pastebin.freeswitch.org/view/707d6998 This is causing mod_cdr_pg_csv to generate 2 CS_DESTROY events instead of 1 but the second one without actually trying the call. Here is the PB for the while call itself. https://pastebin.freeswitch.org/view/0fd51948 Not able to figure out what’s going on. Already filed a Jira FS-9774 just in case. -- Sent with Airmail

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Re: [Freeswitch-users] mod_fifo and early media
by Dominique Jeannerod 25 Nov '16

25 Nov '16

Hi, thanks for your answer. I'm not a developper, but will look at that with someone in my team. What I'm currently planning to do (if possible), and submit as a patch for mod_fifo : Add a parameter to make mod_fifo manage two modes : 1. Normal mode, where the A-Leg gets answered (200 OK). 2. "Early media" mode, where the A-Leg is pre-answered, and kept in early media mode, until the B-Leg answers. The A-leg is then (automatically) answered when the B-Leg answers. Does it make sense ? Dominique Jeannerod Interact-iv.com Infrastructure Manager 2016-11-22 18:52 GMT+01:00 Sergey Safarov <s.safarov(a)gmail.com>: > You can try patch mod_fifo source code and comment > "switch_channel_answer(channel);" strings. > Then you can manage signal is passed to a-leg from dialplan. > > вт, 22 нояб. 2016 г. в 13:24, Dominique Jeannerod > <dominique.jeannerod(a)interact-iv.com>: >> >> Hello, >> >> i'm working on a project where I need to manage a call waiting queue, in >> early media mode, to have a free waiting time, and start billing only after >> the call is really answered. >> >> The general call flow is : >> 1- incoming call to a service number >> 2- early media answer (pre-answer) to manage voice messages >> 3- the call is put in a waiting queue, and the caller ears a message >> (MOH), still in early media >> 4- the queue is associated to a unique destination number, and then one an >> only call at a time is picked-up from the queue and bridged to the >> destination. >> 5- When the call is established (200 OK), the incoming call is also >> answered (200 OK), which starts the billing. >> >> >> Question : this looks like a basic call queue handling, and mod_fifo could >> be a perfect fit for that ... except that mod_fifo answers the call with a >> 200 OK >> >> Is it possible to manage early media with mod_fifo ? >> >> Does someone have a best practice, experience, or advice to share on this >> matter ? >> >> >> Thanks with anticipation >> >> D. Jeannerod >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.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

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[Bridge two channels on talk]
by Bilal Abbasi 25 Nov '16

25 Nov '16

Hi Users, I want to bridge two channels on talk event,scenario is Call comes into FS and put on park. Another call is originated and on talk event i want to bridge that with the parked leg uuid. I will use uuid_brige to bridge two legs, But how can i execute bridge on talk event? i am looking onto some thing "execute on talk" that i can pass with originate command(ideally). P.S: i found on talk event but dont know how to use that, any help is highly appreciated. Regards Abbasi

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How to enable IPv6?
by Charles Yu 25 Nov '16

25 Nov '16

Hi, We want to use Freeswitch at IPv6 environment. How to do it? I saw there's a global variable "local_ip_v6". I think I need to change the default global domain variable ( at vars.xml ) as "local_ip_v6" and other module configs that use global variable "local_ip_v4". Are these right? Thanks in advance, Charles

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Distinctive ringtone on mod_verto like SIP Header Alert-Info
by José Lopes 24 Nov '16

24 Nov '16

Hello, On mod_verto there is any way to forward the SIP Alert-Info Header (distinctive ringtone) to verto client for a new call? Thanks in advance. Best regards, Jose Lopes

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Unable to make audio-only conference
by Kiran Ravuri 23 Nov '16

23 Nov '16

Hi all, I am unable to make audio-only conference call. I am using version 1.6. Even after I choose "wideband" profile, video is negotiated in SDP. Please let me what i should do. BRs, Kiran

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mod_vlc playback hangup issue
by Anonim Stefan 23 Nov '16

23 Nov '16

Hi, I'm trying to playback a .wav using mod_vlc (.wav obtained via maryTTS). Dialplan extension looks like this: <extension name="tts"> <condition field="destination_number" expression="^tts$"> <action application="sleep" data="500"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="log" data="INFO before"/> <action application="playback" data="shout:// 172.16.36.67:59125/process?INPUT_TYPE=TEXT&INPUT_TEXT=${tts_src_data}&LOCALE=en_US&VOICE=cmu-slt-hsmm&OUTPUT_TYPE=AUDIO&AUDIO=WAVE "/> <action application="log" data="INFO after"/> <action application="hangup"/> </condition> </extension> The .wav gets played correctly with [1] Freeswitch log output. The problem is that after the file ends, the channel never hangup(i.e. show channels won't decrease), even after I end call. Actually, I won't receive 200OK for my BYE, as seen in the attached picture. I am using FreeSWITCH version: 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit). Is this a bug ? Thank you, Stefan [1] http://pastebin.com/Q364uHvp

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Bridge an incoming call to an external SIP address
by Cato Gonzalez 23 Nov '16

23 Nov '16

Hi everyone, I am trying to get FS to route inbound calls to an outside sip provider but I am getting the call hung up on the FS side just upon the first or second ring on the B-leg. The diaplan action being executed is: <action application="bridge" data="sofia/internal/ XXX(a)sip.linphone.org"/> I have a softphone registered to this provider sip.linphone.org and it works well when calling XXX from some other IP phone. FS reports this in the logs: [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@48000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/ anonymous(a)webrtc.sip.mydomain.co Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6760 Channel sofia/internal/anonymous(a)webrtc.sip.mydomain.co entering state [terminated][500] [NOTICE] sofia.c:7779 Hangup sofia/internal/anonymous(a)webrtc.sip.mydomain.co [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] [DEBUG] switch_core_codec.c:246 sofia/internal/ anonymous(a)webrtc.sip.mydomain.co Restore previous codec opus:116. [NOTICE] switch_ivr_originate.c:3523 Hangup sofia/external/ XXX(a)sip.linphone.org [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] Result is the same if I try to call any other sip destination. I appreciate any input anyone may give: have been working on this for a week already. Thanks, Cato

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freeswitch sending '481 Call Does Not Exist' in return of BYE
by devang nathwani 23 Nov '16

23 Nov '16

Hello, Here are the sip traces of two legs leg1 UAC -> proxy -> media leg2 media -> proxy -> provider leg1 http://pastebin.com/G0jnF75t leg2 http://pastebin.com/299dZyN4 from leg1, when freeswitch(media) is sending back 480 temporarily unavailable after 183 session progress, UAC is sending ack and bye, now in return freeswitch is sending '481 Call Does Not Exist' my question is why? why freeswitch is sending 481 in return of ack and bye? here, 11.23.16.12 is UAC 11.23.16.13 is Opensips(Proxy) 11.23.16.16 is Freeswitch(Media) Thanks, Devang Nathwani

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Freeswitch routing inbound calls over SIP instead of TLS/SRTP
by Tim Smith 23 Nov '16

23 Nov '16

Debian GNU/Linux 8 (jessie) Linux my 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) I have a Vtech handset with TLS/SRTP enabled registered with Freeswitch as below: Call-ID: a0000a0a000aa000 User: 2001(a)my.example.com Contact: "my" <sips:2001@198.51.100.81:58348> Agent: Vtech Vesa VSP736A 2.0.3.2-0 Status: Registered(TLS)(unknown) EXP(2016-11-18 10:56:57) EXPSECS(3646) Ping-Status: Reachable Ping-Time: 0.00 Host: my IP: 198.51.100.81 Port: 58348 Auth-User: 2001 Auth-Realm: my.example.com MWI-Account: 2001(a)my.example.com sofia_contact is happy : freeswitch@my>sofia_contact internal/2001 sofia/internal/sip:2001@198.51.100.81:58348 I have an inbound dial plan configured as follows: <include> <extension name=“test_inbound"> <condition field="destination_number" expression=“^(15550100)$"> <action application="set" data="domain_name=$${domain}"/> <action application="bridge" data="${sofia_contact(internal/2001)}"/> </condition> </extension> </include> The problem is Freeswitch is sending invites over SIP/RTP and not TLS/SRTP and so the calls never get through : INVITE sip:2001@198.51.100.81:58348 SIP/2.0 Via: SIP/2.0/UDP 203.0.113.4;rport;branch=z9hG4bKvHjgXXpFF77XK From: "Anonymous" <sip:anonymous@203.0.113.4>;tag=rKmXQjZN8SFXp To: <sip:2001@198.51.100.81:58348> m=audio 32190 RTP/AVP 8 98 9 101 a=rtpmap:8 PCMA/8000 a=rtpmap:98 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20

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