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new install Debian 9 and FS 1.8 - Seg Fault
by Sean Devoy 24 Jun '19

24 Jun '19

Hi, I have a brand new install of Debian 9 and FS 1.8 (dist) fs_cli -rRS results in error connecting freeswitch -c ends with: Adding API Function 'opus_debug' Segmentation fault (core dumped). This is my first attempt at FS on a nat'ed PC. I ran Debian install I ran FS dist install. I changed Password. NOTHING else. What did I miss? Thanks, sean

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No outbounds calls
by paolo.visnoviz＠vipcomputers.it 21 Jun '19

21 Jun '19

Ok guys, I give up. So I must ask help. :-( In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please? This is my gateway conf: <include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/>   <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/>  <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/>   </gateway> <gateway name="123456789"> <param name="username" value="123456789(a)ims.my_provider.net"/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> This is my dialplan:  <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/>  <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include> I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209. This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards -- Distinti saluti /Paolo Visnoviz/

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Send conference commands through Verto's RPC client without moderator privileges
by Anton Olofsson 20 Jun '19

20 Jun '19

Is it possible to make users able to execute commands such as muting themselves through the RPC client in Verto without the user needing moderator privileges? I'm trying to do this in order for the user's mute status to be broadcast to all other users through the LiveArray (and not just having the user disable the streams locally).

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Calls between extensions with no audio on NAT
by Jose Figueroa 19 Jun '19

19 Jun '19

Hello, I'm having a problem with a NAT scenario, I have some pcs (192.168.10.102 and 192.168.10.103) using verto behind a NAT calling to a LAN FS (192.168.10.143). I added a gateway and the callout is working good enough, but the calls between extensions I am not getting audio, I just get the "No audio stun for a long time!". I set in many ways the following: <param name="ext-rtp-ip" value="192.168.30.143"/> <param name="ext-sip-ip" value="auto"/> I tried auto, auto-nat, $${external_rtp_ip} and $${external_sip_ip} respectively, and that one I just added with no luck. What can be the problem with this? Thanks Jose

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group_confirm_key=exec : freeswitch start exec on 183-session progress instead of 200-OK..
by Julien Terrasson 19 Jun '19

19 Jun '19

Good morning, I'm using freeswitch Version 1.8.2 -3-a98a958ac3 64bit, and i have an issue with the following scenario : I bridge an outgoing call through a PSTN gateway, using group_confirm_key = exec, mainly to ask the called party confirmation before establishing the call. To do this, i use group_confirm_key=exec to start a lua script that : * Plays a vocal prompt to warn the calleR party that he has to wait for his call to be accepted. * Plays a vocal prompt to request the calleD to press a DTMF to confirm that he accept the call. This worked fine until i start dialing numbers that make the PSTN gateway to return a 183-session progress before 200-OK. When this happen, the group_confirm script is triggered on reception of 183-session progress : before the calleD party actually has answered the call ! Maybe i'm missing something, but I don't get why group_confirm_key=exec start on 183-session progress, as per my understanding, confirmation message should be requested only when B party has answered (200-OK). Does somebody knows how group_confirm_key=exec is supposed to work ? Should this be reported as an issue ? Regards, J.Terrasson

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How to install V8?
by Yehavi Bourvine 19 Jun '19

19 Jun '19

Hello, I am still running version 1.6, since I have a lot of JavaScript scripts which I cal from Freeswitch using mod_v8. In order to upgrade to 1.8 I need to install the V8 library. I've tried on Centos-7 and Ubuntu-18, but the install package is either old or not existent. I've tried installing from google's source code, but the configure script does not find it. How do I install V8? Thanks, __Yehavi:

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Alternate Voicemail Outgoing Message
by Sean Devoy 17 Jun '19

17 Jun '19

Hi, Is there a way to control the outgoing message when directing a call to VM? An Exec wants his VM calls to go to his office manager but with a different outgoing message than her own VM. Thanks, Sean

2 1

SDP BUNDLE support
by Sowmya Venkat 15 Jun '19

15 Jun '19

Hi, I am trying to understand the expected behavior of latest FS when the client sends SDP offer with RTP BUNDLE and uses the exact same IP + port combination for both audio and video media components. Is BUNDLE ( https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-39) supported by FS? If not, can you pls let me know the when it is planned to add the support for the same? I am seeing the below errors in fs_cli. Can you pls confirm if the issue is because BUNDLE is not supported by FS? 2019-06-12 07:01:03.143578 [DEBUG] switch_core_media.c:3236 alternate payload received (received 96, expecting 111) 2019-06-12 07:01:03.143578 [WARNING] switch_core_media.c:3259 Could not change to payload type 96, ignoring... 2019-06-12 06:52:36.463638 [ERR] mod_opus.c:921 Decoder Error: buffer too small fs:4080 plc:false! 2019-06-12 06:52:36.463638 [ERR] switch_core_io.c:633 Codec RAW Signed Linear (16 bit) decoder error! [9] 2019-06-12 06:52:36.543581 [ERR] mod_opus.c:921 Decoder Error: corrupted stream fs:4080 plc:false! Here are the relevant SDP m and a lines from the client's offer: a=group:BUNDLE audio video .... m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 a=mid:audio a=rtcp-mux a=rtpmap:111 opus/48000/2 a=candidate:1 1 udp 2013266431 35.162.237.109 38673 typ host .... m=video 9 UDP/TLS/RTP/SAVPF 96 98 100 102 127 125 108 a=mid:video a=rtcp-mux a=rtpmap:96 VP8/90000 a=candidate:1 1 udp 2013266431 35.162.237.109 38673 typ host Thanks Sowmya

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ICTFax version 4.0 , Open source Fax over Ip software
by Tahir Almas 14 Jun '19

14 Jun '19

We are pleased to announce that ICTFax 4.0 has been released developed over freeswitch , ictcore and angular framework. New features include ATA / Extension support and REST API http://ictfax.org/ictfax-4.0-released-with-ata-api-support Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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DTLS 1.2 support
by Sowmya Venkat 14 Jun '19

14 Jun '19

Hi, Can you please let me know when it is planned to add DTLS Ver 1.2 support in Freeswitch ? Thanks Sowmya

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