February 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Testing package installation on Ubuntu Xenial
by erxspin - 20 Feb '17

20 Feb '17

Hello ! # apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 1FDDF413C2B201E5 Executing: /tmp/tmp.7zbJk1hRSC/gpg.1.sh --keyserver keyserver.ubuntu.com --recv-keys 1FDDF413C2B201E5 gpg: requesting key C2B201E5 from hkp server keyserver.ubuntu.com gpg: key C2B201E5: public key "Freeswitch Commercial Package Repo ( https://repo.freeswitch.com) <wking(a)freeswitch.org>" imported gpg: Total number processed: 1 gpg: imported: 1 (RSA: 1) > I see the xenial directory here: > <http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6/> > http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6/ , but it complains > of not being signed: > # curl > https://files.freeswitch.org/repo/ubuntu/freeswitch-1.6/freeswitch_archive_g > 0.pub | apt-key add - > # echo "deb <http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6/> > http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6/ xenial main" > > /etc/apt/sources.list.d/freeswitch.list > # apt-get update > Get:4 http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6 xenial > InRelease [3,181 B] > Ign:4 http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6 xenial > InRelease > [.] > W: GPG error: http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6 xenial > InRelease: The following signatures couldn't be verified because the public > key is not available: NO_PUBKEY 1FDDF413C2B201E5 > W: The repository 'http://files.freeswitch.org/repo/ubuntu/freeswitch-1.6 > xenial InRelease' is not signed. > (Note: Yes I've been successful with Debian Jesse, not successful with > CentOS 7 because of Flite and V8 issues, but now am trying in Ubuntu as > well) -- wbr, erxspin

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One-way audio when connecting incoming SIP call and WebRTC call using Valet Parking
by Donny Hardyanto 17 Feb '17

17 Feb '17

Hi, I have problem got one-way audio when connection incoming SIP call and WebRTC call using ValetParking. This is my setup: Browser(WebRTC/Opus)->FS Box 1->FS Box 2 (Core)<-SBC(SIP/G711)<-Telco Network<-Handphone. The call came from SBC as SIP call and FS Box 2 accept and park call on valet parking. Browser place call to FS Box 1 and FS Box 1 will bridge to FS Box 2, and in FS Box 2, this call will retrieve call from valet parking. It result Handphone user can hear Browser user, but browser user cannot hear handphone user. How to solve this problem? Thanks, Donny

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FS service suddenly stops, no errors.
by Mimiko 16 Feb '17

16 Feb '17

Hello. Lately I have a problem with FS. It suddenly stops working. In logs there are no errors, even no segfault error. It happens randomly after 2-3 days of working. This problem started after installing postgres and activating cdr log to DB. Where should I dig? -- Mimiko desu.

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iLBC is missing in Deb8 repo
by Volodymyr Fedorov 16 Feb '17

16 Feb '17

Hi someone have problem that now ilbc module is not in Debian packages ? root@Debian-86-jessie-64-minimal:~# apt-cache search freeswitch-mod|grep ilbc root@Debian-86-jessie-64-minimal:~# It was available in 1.16.14 and disappear now in 1.6.15 . My apt policy for other modules: apt-cache policy freeswitch-mod-xml-curl freeswitch-mod-xml-curl: Installed: (none) Candidate: 1.6.15~32~bec4538-1~jessie+1 Version table: 1.6.15~32~bec4538-1~jessie+1 0 500 http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie/main amd64 Packages root@Debian-86-jessie-64-minimal:~# -- Best regards, Volodymyr

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Re: [Freeswitch-users] Do I need a STUN-Server?
by Do Nguyen Ha 15 Feb '17

15 Feb '17

there is no need stun server if Freeswitch in proxy media mode On Feb 16, 2017 00:05, "Thorsten Göllner" <tg-maillistings(a)level5.de> wrote: Hi, I read some docs about STUN/TURN-Servers and I understand the concept behind it (nearly). Do I need a STUN-Server when a SIP-Client (behind NAT) must connect to a Freeswitch-Server (public ip)? Or is a STUN-Server only needed if a SIP/P2P-Connection must be established? Thank in advance, Thorsten _________________________________________________________________________ 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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Centos 7 Install from repo
by Wesley Akio 15 Feb '17

15 Feb '17

Hi all, I'm trying to install freeswitch from the repo( http://files.freeswitch.org/freeswitch-release-1-6.noarch.rpm) using the provided instructions( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7) as I did many times before and today I hit a snag: yum install freeswitch-config-vanilla ... --> Finished Dependency Resolution Error: Package: freeswitch-config-vanilla-1.6.15-1.el7.centos.x86_64 (freeswitch) Requires: freeswitch-codec-ilbc2 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest Trying to force the previous version creates another issue as it picks the latest freeswitch package as a dep and fails: yum install freeswitch-config-vanilla-1.6.14 ... --> Finished Dependency Resolution Error: Package: freeswitch-config-vanilla-1.6.14-1.el7.centos.x86_64 (freeswitch) Requires: freeswitch = 1.6.14-1.el7.centos Available: freeswitch-1.6.0-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.0-1.el7.centos Available: freeswitch-1.6.2-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.2-1.el7.centos Available: freeswitch-1.6.5-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.5-1.el7.centos Available: freeswitch-1.6.6-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.6-1.el7.centos Available: freeswitch-1.6.8-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.8-1.el7.centos Available: freeswitch-1.6.9-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.9-1.el7.centos Available: freeswitch-1.6.10-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.10-1.el7.centos Available: freeswitch-1.6.11-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.11-1.el7.centos Available: freeswitch-1.6.12-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.12-1.el7.centos Available: freeswitch-1.6.13-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.13-1.el7.centos Available: freeswitch-1.6.14-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.14-1.el7.centos Installing: freeswitch-1.6.15-1.el7.centos.x86_64 (freeswitch) freeswitch = 1.6.15-1.el7.centos You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest I tried a fair amount of combinations and install orders to no avail. Do you guys have any idea on an workaround besides installing the RPMs manually? Cheers, Wes -- <http://www.tuntscorp.com/> *Wesley Akio* Travessa da Lapa 96 - Sala 24 Centro – Curitiba – 80010-190 : +55 (41) 9950-0033 : +1 (305) 842-7095 : wesley_akio_

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Re: [Freeswitch-users] one way speech path after attempting voice path redirecting in new invite
by Bejan, Alex 15 Feb '17

15 Feb '17

Hello, This is my first posting into this list although I monitored activities in the past. Freeswitch behaves great in our environment - thank you all who contribute! We started recently observing some one way speech path issues in the following scenario and I would like to ask for suggestions, please: We have incoming calls from provider to FS that further bridge to different Standard SIP PBX. Call is established to SIP device connected on the PBX. The PBX immediately invites back attempting to redirect rtp path to start recording the call. FS starts sending RTP to the new destination. There are 2 different very similar PBX environments connected to the same FS: - In one the PBX starts sending RTP even before the connection is ACKed. Recording succeeds. - In the second one, PBX starts sending RTP almost one second later. It seems FS reverts back to the original speech path, while the other devices remain unaware, resulting in one way speech path. The rtp-timeout-sec and rtp-hold-timeout-sec are on the default values  <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> Are there any other parameters that could have such an effect? Your suggestions on further troubleshooting or how to make this work even when RTP is not being received will be greatly appreciated! Thank you very much! Regards

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Do I need a STUN-Server?
by Thorsten Göllner 15 Feb '17

15 Feb '17

Hi, I read some docs about STUN/TURN-Servers and I understand the concept behind it (nearly). Do I need a STUN-Server when a SIP-Client (behind NAT) must connect to a Freeswitch-Server (public ip)? Or is a STUN-Server only needed if a SIP/P2P-Connection must be established? Thank in advance, Thorsten

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Voicemail Forward in MultiDomain setup
by Jai Rangi 15 Feb '17

15 Feb '17

We are using Freeswitch for a multidomain setup and experiencing problem in forwarding voice mail message to another extension in the same domain. This is what I see in the logs 2017-01-31 21:23:37.116117 [WARNING] mod_voicemail.c:3268 Cannot locate domain company.phonesystemdomain.com <http://j2h.la.cebodtelecom.com> 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.116117 [ERR] mod_voicemail.c:1768 Failed to forward message to 130 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.116117 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 30a4bcaa-cd5a-4be8-88bf-94e888f98a16 2017-01-31 21:23:37.136121 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-that_was_an_invalid_ext.wav] (en:en) Anyone else experiencing this, any fix? Not sure if this is a bug. I have experienced this in 1.4, 1.6.6, 1.6.9 and latest stable 1.6.13. Thank you, Jai

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Use inband dtmf only
by Nikolay Zaytsev 15 Feb '17

15 Feb '17

Hello, I have a problem with an inbound provider: sometimes dtmfs do not reach the freeswitch. Provider told me that occurred because I am using G711 with rfc2833, and I should use only G711. Inbound call from the carrier to FreeSwitch: SDP in the INVITE carrier --> FreeSwitch v=0 o=Sansay-VSXi 188 1 IN IP4 x.x.x.x s=Session Controller p=+X XXXXXXXX c=IN IP4 x.x.x.x t=0 0 m=audio 17774 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 SDP in the 200 OK message from FreeSwitch-->carrier: v=0 o=FreeSWITCH 1486794789 1486794790 IN IP4 x.x.x.x s=FreeSWITCH c=IN IP4 x.x.x.x t=0 0 m=audio 30088 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Carrier wants me to get rid of those session attributes in my 200 OK message: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 and media to be like this m=audio 30088 RTP/AVP 0. Could you please advise me how to do it? I have already tried to set up dtmf-type=none and liberal-dtmf=false in SIP profiles, dtmf_type=none in Dialplan. In addition I tried start_dtmf before application answer. Also in the vars.xml I specified to use PCMU and PCMA only for both global_codec_prefs and outbound_codec_prefs. My dialplan instruction for inbound call: <extension name="inbound_call" continue="false"> <condition field="destination_number" expression="^79" > <action application="answer"/> <action application="sleep" data="1000"/> <action application="socket" data="127.0.0.1:8800 async full"/> </condition> </extension> Im using several FreeSwitch boxes, one of them is 1.6.2 version another one is 1.6.14, but this problem occurres on both. Thank you for your time and best regards, Nikolay Zaytsev

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