October 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

No RTP after unholding call
by sagar malam 09 Oct '18

09 Oct '18

Caller A and Caller B are on call.Caller A puts call on hold for 2 mins.During these 2 mins FS does not send any RTP to firewall due to which firewall closes RTP port of caller A due to udp inactivity can we configure FS to send something like empty RTP to caller A ? I did not find any settings in wiki Thanks in advance

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Codec negotiation issue
by sagar malam 08 Oct '18

08 Oct '18

I am facing an issue related to codec negotiation.Let me explain scenario inbound/outbound codec pref list : g722,g729,g711 late negotiation enabled mix inbound outbound codec is enabled When call hits FS, i give instant ringback to call which mean g722 codec will be negotiated for early media Now when callee answers call with g729 FS does transcoding from g722: g729 Instead it should select g729 on both legs and tell aleg to update codec from g722 to g729 to avoid transcoding.

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execute_on_answer does not work when group dial is used for bridging to multiple extensions
by Jenish Patel 06 Oct '18

06 Oct '18

Hi everyone, I am trying to use the bind_digit_action on the b-leg , and as wiki suggested I am using execute_on_answer to assign the bind_digit on the b-leg. This works like charm in case of the normal simple bridge. But in case of the Ring-group like bridge when I bridge the multiple users at the same time, it does not get executed on the channel answer. I see in the logs that in working case when b-leg answered , the execute_on_answer gets executed. But in ring group case it does not get executed at all. Am I missing something here ? Like is there any other param needs to be set to use the execute_on_answer in the ring group like structure. And if it is not possible, can you suggest me any other way to use bind_digit on b-leg from dialplan itself. Below is my dialplan for ring group. <include> <extension name="Ring_all"> <condition field="destination_number" expression="3301"> <action application="export" data="dialed_extension=$1"/>   <action application="bind_digit_action" data="features,*1,exec:valet_park::my_lot auto in 8001 8999,both,self"/> <action application="export" data="nolocal:execute_on_answer_11=features,*1,exec:valet_park::my_lot auto in 8001 8999,both,self"/> <action application="bind_digit_action" data="features,*2,exec:execute_extension,START_RECORDING XML features,both,self"/> <action application="export" data="nolocal:execute_on_answer_12=features,*2,exec:execute_extension,START_RECORDING XML features,both,self"/> <action application="bind_digit_action" data="features,*3,exec:execute_extension,att_xfer XML features,both,self"/> <action application="export" data="nolocal:execute_on_answer_13=bind_digit_action features,*3,exec:execute_extension,att_xfer XML features,both,self"/> <action application="bind_digit_action" data="features,*4,exec:execute_extension,blind_xfer XML features,both,self"/> <action application="export" data="nolocal:execute_on_answer_14=bind_digit_action features,*4,exec:execute_extension,blind_xfer XML features,both,self"/> <action application="bind_digit_action" data="features,*51,exec:execute_extension,START_RECORDING XML features,aleg,self"/> <action application="export" data="nolocal:execute_on_answer_15=features,*51,exec:execute_extension,START_RECORDING XML features,aleg,self"/> <action application="bind_digit_action" data="features,*52,exec:execute_extension,STOP_RECORDING XML features,aleg,self"/> <action application="export" data="nolocal:execute_on_answer_16=features,*52,exec:execute_extension,STOP_RECORDING XML features,aleg,self"/>   <action application="set" data="ringback=/home/jenish/moh.wav"/> <action application="set" data="transfer_ringback=/home/jenish.moh.wav"/> <action application="set" data="call_timeout=300"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="REC_TYPE=GROUP"/> <action application="set" data="REC_STATUS=Y"/> <action application="export" data="effective_caller_id_name=Department_group_call"/> <action application="record_session" data="/usr/local/freeswitch/recordings/1000/GROUP/${strftime(%Y-%m-%d-%H-%M-%S)}_from_${caller_id_number}_to_${destination_number}__${uuid}.wav"/> <action application="set" data="REC_PATH=/usr/local/freeswitch/recordings/1000/GROUP/${strftime(%Y-%m-%d-%H-%M-%S)}_from_${caller_id_number}_to_${destination_number}__${uuid}.wav"/> <action application="set" data="export_vars=call_direction,original_source,REC_PATH,REC_TYPE,REC_STATUS"/> <action application="bridge" data="[absolute_codec_string=^^:PCMA@8000h@20i@64000b ]${sofia_contact(internal/1000101@ ${domain_name}},[absolute_codec_string=^^:PCMU@8000h@20i@64000b ]${sofia_contact(internal/1000100@ ${domain_name}},[absolute_codec_string=^^:PCMA@8000h@20i@64000b:PCMU@8000h @20i@64000b]${sofia_contact(internal/1000102@ ${domain_name}},[absolute_codec_string=^^:PCMA@8000h@20i@64000b:PCMU@8000h @20i@64000b]${sofia_contact(internal/1000103@${domain_name}}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> </include> -- Thanks & Regards Jenish Patel +91 9033987576

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Wrong response SIP port
by morfair＠gmail.com 05 Oct '18

05 Oct '18

I have external profile on port 5080. When FS get INVITE on 5080 from remote 5060 port, FS response on 5080 port too instead 5060 20.20.20.20 - fs 10.10.10.10 - remote sip 18:23:49.675349 IP 10.10.10.10.5060 > 20.20.20.20.5080: SIP: INVITE sip:7001@20.20.20.20:5080;user=phone SIP/2.0 18:23:49.678953 IP 20.20.20.20.5080 > 10.10.10.10.5080: UDP, length 363 How to fix whis? FS should response on originate udp port.

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Multiple concurrent Verto calls in browser
by Michael Avers 05 Oct '18

05 Oct '18

Hello, I'm using Verto Communicator as a base for building a simple app to make/receive calls - and it works great. I'm trying to add the ability to manage multiple concurrent calls. My first thought is to create a new Verto service (in Angular) that maintains call state and reference to currently active call and otherwise keeps all calls in an array. Then to switch calls it'd place the currently active one on hold and then unhold any of the other calls in the array. Does this make sense? Any tips on handling multiple calls with Verto? Do I need a separate websocket connection for each? (they're all using the same proxy and credentials). Thanks! Mike

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Re: [Freeswitch-users] npa_nxx_company_ocn database?
by Avi Marcus 04 Oct '18

04 Oct '18

Ah, it was linked from mod_cidlookup page: http://files.freeswitch.org/npa-nxx-companytype-ocn.csv But `curl --head` tells me: Last-Modified: Wed, 08 Jul 2009 01:19:32 GMT Seems my carriers only provide intra or interstate, no LATA calculations necessary. Anyone have an updated npa/nxx per state list? Do any npa's cross state lines? Perhaps I only need the much smaller NPA list. According to that 2009 data, none do. -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel) 020-3298-2875 (UK) On Wed, Oct 3, 2018 at 9:04 PM Avi Marcus <avi(a)avimarcus.net> wrote: > mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state > or inter/intra-lata calls. > > Is there an updated database somewhere? I have npa_nxx_company_ocn from > someone from years ago. > > Thanks! > > -Avi Marcus > 1-718-989-9485 (USA) > 02-372-1570 (Israel) > 020-3298-2875 (UK) >

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Astricon: OpenSource Soiree
by Giovanni Maruzzelli 04 Oct '18

04 Oct '18

Hello friends and collegues, A nice get together has been organized for evening Wednesday 10th at 7pm The idea is to have drinks and chitchat about opensource telephony and webrtc, and all opensource project founders, members, and practitioners are invited. Please join me, as distinguished participants too! Check it out at http://party.officering.com (no animals has been harmed in this mail) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Call Group Assistance
by Lee Barnard 04 Oct '18

04 Oct '18

Hi Guys, Having an issue with a call group. Issue: Have 3 users 1. 1 External 2. 2 Internal Call rings to all 3 endpoints, however, when rejecting call on the external entity it clears call on all legs. If call is rejected on either internal entities continues to ring on all other entities. Can anyone give a pointer as to what to try to get the functionality the same as the internals across the external entities. NOTE: You can see some artefacts of things tried in the dial-string. Configuration is built from LUA script and appears in debug as below: 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Call Group Request 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2000 and Domain: domain.co.uk 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result: 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Extension: 2001 and Domain: domain.co.uk 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Query Result: 123456789 2018-10-03 12:32:52.430641 [NOTICE] switch_cpp.cpp:1365 Action Huntgroup Returns: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="domain.co.uk"> <groups> <group name="groupname"> <users> <user id="123456789"> <params> <param name="dial-string" value="{hangup_after_bridge=true,fail_on_single_reject=false}sofia/internal/123456789(a)domain.co.uk"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> <user id="2000" type="pointer"/> <user id="2001" type="pointer"/> </users> </group> </groups> </domain> </section> </document> 2018-10-03 12:32:52.430641 [DEBUG] freeswitch_lua.cpp:382 DBH handle 0xc6bb90 released. 2ffa1d17-2ae3-4920-8f48-23798644ed25 EXECUTE sofia/internal/987654321@IPADDRESS bridge([hangup_after_bridge=true,fail_on_single_reject=false]sofia/internal/123456789@domain.co.uk,user/2000@domain.co.uk,user/2001@domain.co.uk<mailto:sofia/internal/123456789@domain.co.uk,user/2000@domain.co.uk,user/2001@domain.co.uk>) Happy to share any additional information required – please be gentle, I am fairly newish to freeswitch. Thanks in advance Lee Kind regards, Lee Barnard Voice Systems Engineer T: 01202612000 F: 01202612111 E: Lee.barnard(a)voip-unlimited.net Online ordering now live! Visit: portal.voip-unlimited.net/ordering<https://portal.voip-unlimited.net/ordering> or speak to your Account Manager today. ________________________________ [VoiP Unlimited]<http://www.voip-unlimited.net> [Force India] <http://www.forceindiaf1.com/partners/voip-unlimited> [ITSPA] <https://www.voip-unlimited.net/voip-unlimited-named-best-business-itsp-fina…> [Facebook]<https://www.facebook.com/pages/VoIP-Unlimited/286623001407365>[cid:image8dda1d.PNG@aaf8f1d3.4fabb9b6]<https://plus.google.com/114885723213491901544>[cid:imagec180ba.PNG@3cf76013.479eeb80]<https://www.linkedin.com/company/voip-unlimited>[cid:image418cf6.PNG@78887a23.4285e855]<https://twitter.com/voipunlimited>[cid:image3d111d.PNG@9a6f9dd3.4dac0962]<https://www.youtube.com/watch?v=TmD1Eg3uCwY> [VU Guard DDoS Mitigation]<http://www.voip-unlimited.net/services/channel-ddos> 6 Albany Business Park, Cabot Lane Poole, Dorset, UK, BH17 7BX. This email including any attachments is intended only for the addressee named above and it may contain confidential or privileged information. If you are not the intended recipient please notify the sender and note that the contents must not be disclosed to anyone else. Voip Unlimited will not be liable for any error in transmission. You should carry out your own virus checks before opening any attachments. Opinions, conclusions and other information that do not relate to the official business of Voip Unlimited are neither given nor endorsed by it.

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npa_nxx_company_ocn database?
by Avi Marcus 03 Oct '18

03 Oct '18

mod_lcr mentions using npa_nxx_company_ocn to determine inter/intra-state or inter/intra-lata calls. Is there an updated database somewhere? I have npa_nxx_company_ocn from someone from years ago. Thanks! -Avi Marcus 1-718-989-9485 (USA) 02-372-1570 (Israel) 020-3298-2875 (UK)

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keep-alive RE-INVITE dropped calls
by Bipin Patel 03 Oct '18

03 Oct '18

hi, im suffering call drops from carrier in roughly 15mins and i reported this to the carrier and they replied with the below: If the duration of the call was a multiple of 15 minutes, please make sure that you can properly respond to the keep-alive RE-INVITE that we sends every 15 minutes. i have no clue on how this works in FS so can any1 point me the right direction on how to sole this, we use FS to route calls from our switch to carrier -- Regards, Bipin

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