July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
by Nathan Neulinger 14 Jul '17

14 Jul '17

I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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macOS FreeSwitch Installer (macFI) Update
by Mario G 14 Jul '17

14 Jul '17

For the macOS folks: macFI <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installati…> was extensively updated and if you use it you should download the latest version. Auto detemines the latest FS stable release, no need to update for new releases. Can now download stable production, master development, and the current branch if needed Other changes, see history <https://freeswitch.org/confluence/display/FREESWITCH/macOS+macFI+Installati…>.

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Error sending queries to pgsql
by Antonio Silva 14 Jul '17

14 Jul '17

Hi all, i use pgsql in core and from time to time i see critical messages like fail to send query, example: [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server closed the connection unexpectedly This was recently, i did an update to current master the previous version was from April, not sure if it could be an error on FS o some other issue on my box.. PG is installed on the same server and the only thing i see from pg is "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed on the same server, running on /dev/shm with the same prio as FS and the process never stopped. anyone has experience this error before? any idea what it could be the cause? Thanks, -- Saludos / Regards / Cumprimentos, António silva

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xml_radius only on bleg
by Jospeh Waite 14 Jul '17

14 Jul '17

Hi Guys I am trying to find a variable available to xml_radius to put in the regex section at the bottom of the config to only send a Star/Stop Radius accounting message for a successful connected b-leg. I don’t want it to send any messages for the a-leg as Im only interested in accounting for CDR’s that actually connect. Any help would be greatly appreciated. Regards

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Manipulating variables & doing a for loop in dialplan
by Joseph Waite 13 Jul '17

13 Jul '17

Hi Guys Been googling but can’t really find anything on how to do what I need. I receive the following in a variable sip_redirect_contact_<index> <sip:441554555666@185.35.228.48:5060>;src_number=441554333444;q=1.00 I need to pull out the 441554333444 after src_number= and before th ; How would I achieve this? Also is there any way to do a looping dial plan, that will execute a section of dial plan for each <index> So I would have a block of xml that would run for each <index> from the sip_redirect_contact. So for each sip_redirect_contact would execute a block of xml for each result until the bridge application connected a b-leg Regards

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sip to webrtc - sdp invalid description
by David Villasmil 13 Jul '17

13 Jul '17

Hello guys, I have this setup: Zoiper-->Kamailio->fs->kamailio->webrtc client(s) Whenever webrtc clients call each other, calls are ok. But when the zoiper (regular sip/tcp) calls, the browsers complaint about: "no ice-ufrag" (firefox) or "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." (Chrome). I am setting the sdp in freeswitch as: <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=false"/> <action application="export" data="zrtp_secure_media=false"/> <action application="export" data="nolocal:rtp_secure_media=true"/> and the actual sdp is: v=0 o=FreeSWITCH 1499354716 1499354717 IN IP4 1.2.3.4 s=FreeSWITCH c=IN IP4 1.2.3.4 t=0 0 m=audio 40954 RTP/SAVP 8 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=crypto:1 AEAD_AES_256_GCM_8 inline:9XidL0Z5VYb0L5CegRZaYVrVfjA0Im Wgu5WyLK0vtg60RTk6Koe8c0sRkjU a=crypto:2 AEAD_AES_128_GCM_8 inline:fLKE1lxhRoVw+D5NVoKFFw06I0Xok/9KbRystQ a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:w/yDN+ ETPuCOiOIxjFLRjbFbDxp2xaxhXz4QVwBXWxJw/GigOURGw8EMv9fVUg a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:evU8MzAtiSHwKb95s4V9IAMpmok06k W9ZGDgH3/Lc3ZytVn2SR4 a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:KKDkT0DssohSeKFsX6tbixRhwYdiIh E6r3u5CCVA a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:D3JgGuOlIxXfHGdqf7lKWqNDAIiJrb OqOKb+erlhPQtBKF4wzomjbN0sBIiE4w a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:c12ebWWzZ1cqZN0v5C5uYzdvtfnw6A ARU3+jGA0WzTSDlDd20vI a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:nPZku7S6hlz2OPAff8T9I8sNFzuZzi Ng64KuvfNS a=crypto:9 AES_CM_128_NULL_AUTH inline:9MNdj7xaingoGY14NUg8iS3dhTqD0X W8FAOLNtmP a=ptime:20 a=nortpproxy:yes I'm pretty new to the webrtc stuff, so any help is greatly appreciated! Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 <+34%20669%2044%2083%2037> ᐧ

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FreeSWITCH in debian stretch (9.0)
by Joel Serrano 13 Jul '17

13 Jul '17

Hi all, Does https://freeswitch.org/jira/browse/FS-9785 being fixed in v1.6.19 mean that FS can now work on debian9? Thanks, Joel.

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FreeSWITCH is Hiring a Software Developer!
by Anthony Minessale 13 Jul '17

13 Jul '17

Apply today: https://freeswitch-solutions-llc.workable.com/jobs/523308 -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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NDLB-force-rport question
by Igor Olhovskiy 12 Jul '17

12 Jul '17

Hi! I’m having strange issues on registering phones behind NAT on FreeSwitch. Schema is Phone (port 5063) -> Router (NAT) (port 1023) -> Freeswitch Phone registering from port 5063. NAT is changing this port to 1023. Freeswitch receiving register message from port 1023, but still replying to port, that listed in Contact Header. So, Router can’t route this request back correctly. NDLB-force-rport and aggressive-nat-detection is enabled on internal profile. What I got, this issue is when router changes port to 1024 or lower. If port is higher that 1024, than it’s working ok. Also this problem with UDP only, with TCP all is ok. Maybe I’m missing something in config? Thanks. Regards, Igor

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Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
by Jonathan Hunter 12 Jul '17

12 Jul '17

Hi Dragos, Thank you for the fast reply! You are correct in terms of call flow. I am originating from a WebRTC client on Janus, which then sends a SIP invite into FreeSWITCH , I then reply to that to set it to 8K, narrowband, this works perfectly ! 😊 However I then bridge the call out to a carrier, and I want to offer in the SDP the maxplaybackrate=8000, however I cant seem to do this, and this is with and without late-negotiation enabled. Am I missing something or do you have any debug suggestions for me please? Many thanks Jon ---------- Forwarded message ---------- From: Oancea, Dragos <dragos.oancea(a)vonage.com<mailto:dragos.oancea@vonage.com>> Date: Wed, Jul 12, 2017 at 1:14 PM Subject: Re: [Freeswitch-users] Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Hi Jon, If both sides follow the RFC , then you should be able to get narrowband end to end. How do you originate the call ? Sounds like you originate a call from a device and then you bridge to the callee, but you have inbound-late-negotiation on, so the OPUS config XML is not loaded, hence fmtp cannot be changed according to config. Regards, Dragos On Wed, Jul 12, 2017 at 1:04 PM, Jonathan Hunter <hunterj91(a)hotmail.com<mailto:hunterj91@hotmail.com>> wrote: Hi Guys, We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ. This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the fmtp maxplaybackrate=8000 paramater. As when FreeSWITCH answers I see; Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10 However when I send an Invite offer out I see; Media Attribute (a): rtpmap:111 opus/48000/2 Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1 The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to; <param name="use-vbr" value="1"/> <param name="use-dtx" value="0"/> <param name="complexity" value="10"/> <param name="maxaveragebitrate" value="14400"/> <param name="maxplaybackrate" value="8000"/> <param name="packet-loss-percent" value="15"/> <param name="keep-fec-enabled" value="1"/> <param name="use-jb-lookahead" value="1"/> <param name="advertise-useinbandfec" value="1"/> And I also set the SIP profile codec list and absolute codec offer to OPUS@8000h, and as I say works great when we answer an offer, but not when we originate. Am I misconfiguring or is this not possible? I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as; maxplaybackrate=8000 Is this possible? I just want to ensure I get narrowband end to end. Many thanks Jon _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@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<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Jonathan Hunter Technical Director /Telephony Developer M:(+44) 7917 190 438 Email:jhunter@voxboxcoms.co.uk<mailto:Email%3Ajhunter@voxboxcoms.co.uk>

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