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VP9 Error in Version 1.8.2
by Joseph Barrero 10 Oct '18

10 Oct '18

I'm receiving the following error message in freeswitch.log ever since updating to version 1.8.2+git~20180926T175525Z~a98a958ac3~64bit (git a98a958 2018-09-26 17:55:25Z 64bit) on Debian 8.11 (amd-64): [ERR] switch_vpx.c:529 VPX encoder WebM Project VP9 Encoder v1.6.0 codec init error: [8:Profile > 1 not supported in this build configuration] The error is present when I'm using WebRTC to connect to FreeSWITCH on either Chrome or Firefox. Any advice on how to fix is greatly appreciated. - Joe

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SNMP v3
by Jerry Chinn 09 Oct '18

09 Oct '18

Anyone know if SNMP v3 is supported in FS 1.6.20? I can't find anything recent on SNMP. Mod_snmp documentation is from 2005. Jerry Chinn Telecom VoIP Specialist NAVIS More Performance. More Profit. tel 541-330-3562 www.TheNavisWay.com<http://www.thenavisway.com/> Facebook<https://www.facebook.com/theNAVISway/> | Twitter<https://twitter.com/NAVISway> | LinkedIn<https://www.linkedin.com/company/navisway> | Blog<https://www.thenavisway.com/blog>

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*.tmp.xml.fsxml Files
by Markus Bönke 09 Oct '18

09 Oct '18

Hallo All, I found an old Post discussing <uuid>.tmp.xml.fsxml files in /var/log/freeswitch.log http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.… <http://lists.freeswitch.org/pipermail/freeswitch-users/2016-October/122997.…> Unfortunately there is no real answer on if they can be disabled or why they are written. We have one server on Version 1.6.20-37-987c9b9~64bit which is writing about 30+ of those files per minute. So maybe something is going wrong here. Are there any further Details on those files available? Thanks and regards Markus

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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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