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Re: [Freeswitch-users] How to reduce default sleep between transfers?
by Ahmed habiba 08 Aug '17

08 Aug '17

As mentioned by Brian you need to change the default password of the extensions in the "freeswitch/conf/vars.xml” as below: <X-PRE-PROCESS cmd="set" data=“default_password=xyzabc”/> *Note xyzabc is just example > > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] How to reduce default sleep between transfers? > Date: August 8, 2017 at 7:04:00 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > That sleep is there on purpose in the configs and has you looked at fs_cli it complains that you’ve not changed the default password! CRIT messages:) > > > > On Tue, Aug 8, 2017 at 10:24 AM Jose David Jurado Alonso <josedavid(a)zennio.com <mailto:josedavid@zennio.com>> wrote: > Hi, > > I'm not sure but you can try to modify some of this timeouts: > > /etc/freeswitch# grep -r "10000" * > ... > dialplan/default.xml: <action application="sleep" data="10000"/> > ivr_menus/new_demo_ivr.xml: timeout="10000" > ivr_menus/demo_ivr.xml: timeout="10000" > skinny_profiles/internal.xml: <param name="digit-timeout" value="10000"/> > ... > > Probably, the correct variable to be modified is in "default.xml". > > > 2017-08-07 14:34 GMT+02:00 Euan Millar <euan(a)ensemblepourladifference.org <mailto:euan@ensemblepourladifference.org>>: > Hi, > > I am quite new to Freeswitch so this is probably a simple config question... > > When transferring a call I notice a 10 second sleep which can make users think that something has gone wrong. It is slightly unnatural between my IVR applications. > > I see this in the log: > > EXECUTE sofia/internal/1003@192.168.0.4:5060 <http://1003@192.168.0.4:5060/> sleep(10000) > > Is there a way to reduce this value? > > Many thanks, > > Euan > > > _________________________________________________________________________ > 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/> > -- > 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/> > Book a phone call (CST) <https://freeswitch.com/appointment> > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch <https://www.gofundme.com/allison-prompts-for-freeswitch> > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > >

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How to reduce default sleep between transfers?
by Euan Millar 08 Aug '17

08 Aug '17

Hi, I am quite new to Freeswitch so this is probably a simple config question... When transferring a call I notice a 10 second sleep which can make users think that something has gone wrong. It is slightly unnatural between my IVR applications. I see this in the log: EXECUTE sofia/internal/1003@192.168.0.4:5060 sleep(10000) Is there a way to reduce this value? Many thanks, Euan

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att_xfer source leg and lua dialplan
by animal_alpha 08 Aug '17

08 Aug '17

Hello everybody! I am using Freeswitch with mod_lua and have the following setup: User 1004 is calling user 101, user 101 does an att transfer to 103, but user 103 see the caller-id of 1004(basically 1004 is calling both 101 and 103: mod_dialplan_xml.c:637 Processing 1004 <1004>->103 in context 1_local_103) 101, not 1004 should call 103 How can i get caller-id of 101 on 103(set -bleg on transfer?? )? 101 user's context is : <context name="1_local_101"> <extension name="101"> <condition field="destination_number" expression="^(101)$"> <action application="bind_meta_app" data="# b s execute_extension::att_xfer XML features"/> <action application="bridge" data="{sip_invite_domain=1.sips.local}sofia/internal/101(a)1.sips.local"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> 103 user's context: <context name="1_local_103"> <extension name="103"> <condition field="destination_number" expression="^(103)$"> <action application="export" data="ignore_display_updates=true"/> <action application="bind_meta_app" data="# b s execute_extension::att_xfer XML features"/> <action application="bridge" data="{sip_invite_domain=1.sips.local}sofia/internal/103(a)1.sips.local"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> features context: <context name="features"> <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="att_xfer" data="{sip_invite_domain=1.sips.local}sofia/internal/${digits}@$${domain}"/> </condition> </extension> <extension name="empty"> <condition field="destination_number" expression="^(.*)$"> <action application="hangup" data="NO_ROUTE_DESTINATION"/> </condition> </extension> </context> -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/att-xfer-source-leg-and-lua-d… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FS AWS does not call between mobile phones connected to the same WiFi
by Jose David Jurado Alonso 07 Aug '17

07 Aug '17

Hi, I have an FS server running on an AWS instance and it does not call between mobiles connected to the same WiFi. The call is successful if each of the mobiles is connected to different Wifi or by mobile network (4G). Obviously FS is always in a different network and I use a domain/URL to routing the calls. I also do not see anything strange in the logs. I think that call does not occur because the server does not find the destination number as registered and hangup/abandoned the call by timeout. It always works except when the two mobiles are connected to the same wifi. My current configuration is the next: vars.xml: <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:mydomain.us"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:mydomain.us"/>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> internal.xml: <param name="apply-nat-acl" value="rfc1918.auto"/> <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> internal-ipv6.xml:    external.xml: <param name="local-network-acl" value="rfc1918.auto"/> <param name="ext-rtp-ip" value="autonat:$${external_rtp_ip}"/> <param name="ext-sip-ip" value="autonat:$${external_sip_ip}"/> external-ipv6.xml: <param name="local-network-acl" value="rfc1918.auto"/>    One call example is "sip:1009@mydomain.us:5080"

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World network tones in TGML
by Allan Kristensen 06 Aug '17

06 Aug '17

Hello, Does anyone have/need a complete database of network tones in TGML ? These: https://www.itu.int/dms_pub/itu-t/opb/sp/T-SP-E.180-2010-PDF-E.pdf I'm thinking a lot of WebRTC clients doesn't have any built in support for generating network tones and must have these served as early media? (at least the ones I've worked with ;-). We need it for different reasons, but before we start building it, I just wanted to check if it was already existing out there? Best Regards, Allan Kristensen

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DNS SRV failover
by Sergey Safarov 06 Aug '17

06 Aug '17

I configured two DNS SRV like "_sip._udp.example.com" and pointed to two FS servers. Than i originate call from third FS server and use "fs_path=sip:example.com" Looks like FS 1.6.18 not detect fail of one server and always routes to only one server of two in SRV records. Is this bug and need to report or i do some thing wrong?

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Httpapi and xml dialplan
by Gregor Nanger 06 Aug '17

06 Aug '17

Can someone please advice me. Would like to develop something with httpapi and need to know about concept. Can I mix httpapi and xml dialplan plan? For example, can I execute answer, call httpapi for some work and then proceed with xml dialplan or httpapi continuous loop for asking what's next? Thank you, Gregor -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si

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TURN Configuration under CGN (RFC6598)
by Nandy Dagondon 05 Aug '17

05 Aug '17

Hi folks, I just learned that our Internet provider implemented CGN when I'm about to experiment WebRTC/mod_verto. I already signed up a free TURN server account at http://numb.viagenie.ca/ to proceed. Where do I place turn: numb.viagenie.ca in the config files? Is this correct? sip_profiles/external.xml: <param name="ext-rtp-ip" value="turn:numb.viagenie.ca"/> <param name="ext-sip-ip" value="turn:numb.viagenie.ca"/> /Nandy

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SIM7100 LTE modem with USB audio
by Stanislav Sinyagin 05 Aug '17

05 Aug '17

Simcom has recently released a new 4G/LTE modem, and it has USB audio support. You can find sim7100_usb_audio_application_note_v0.01.pdf with details at the vendor site, or at techship.com after registration. It transmits 8kHZ, 16-bit PCM audio via a USB UART simulation. So, in theory, gsmopen module may be adapted to it (or maybe a new module is worth starting). I ordered a sample, will check it out soon. cheers, stanislav

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ASR with Google speech API
by Murugan Pandian 04 Aug '17

04 Aug '17

Hi, Is there any way i can use Google Speech API with freeswitch ASR Regard's Murugan Pandian

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