November 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

gateway max-calls
by Tihomir Culjaga 05 Nov '17

05 Nov '17

hi, is there any way to limit the number of channels going through a gateway ? i have a scenario like this: bgapi originate [origination_uuid=uuid1]user/1001 &park() bgapi originate [origination_uuid=uuid2]sofia/gateway/gw1/1234 uuid_bridge uuid1 uuid2 since im not hitting the dialplan here, i cannot use mod_limit .. or can i ? how to set limits on a specific gateway in this scenario ? thanks, Tihomir.

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Re: [Freeswitch-users] Flasher
by Kim Culhan 05 Nov '17

05 Nov '17

On Sun, November 5, 2017 7:35 am, Rick Jarvis wrote: > Looking to build a bright flashing light that flashes when an extension is > ringing. Has anyone done anything like this? Take a look at: https://www.vikingelectronics.com/product_docs/product_manuals/VE__SR-IP__S… You might need to adapt the above item to provide for the 'bright flashing light' requirement. -kim

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Gateway Failover
by Gregor Nanger 05 Nov '17

05 Nov '17

Hi, need some advice. Is it possible to set variable on failover scenario? I have failover in bridge /gateway1|/gateway2, but would like to set variable on a leg when failover occurs. Something like: /gateway1/|{myvariable=2}/gateway2 Is this possible? Thank you, Gregor

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Double DSP payload?
by Luis Daniel Lucio Quiroz 03 Nov '17

03 Nov '17

Hi Is there a reason why freeswitch sends a double DSP payload in the SIP invite from a gateway? The problem is that the payload is too big that It gets fragmented and then it fails because of that. v=0 o=FreeSWITCH 1509455513 1509455514 IN IP4 208.90.97.6 s=FreeSWITCH c=IN IP4 208.90.97.6 t=0 0 m=audio 16724 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp-mux a=rtcp:16724 IN IP4 208.90.97.6 a=ptime:30 m=audio 16724 RTP/AVP 18 0 8 3 101 13 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtcp-mux a=rtcp:16724 IN IP4 2 Any comments? -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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Query a Database
by Social Boh 03 Nov '17

03 Nov '17

Hello List, which is the best way to query a Database and use the results in Dialplan? Thank you. Regards -- --- I'm SoCIaL, MayBe

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Call send from Freeswitch to Asterisk
by Uday kumar 03 Nov '17

03 Nov '17

Hello All, I am using Freeswitch on Debain 8. Configured few DIDs. Calls coming from provider and I am trying to send Call to a Asterisk server but getting an error "MANDATORY_IE_MISSING". DID Provider (3.3.3.3) => FS (192.168.1.70) => Asterisk (192.168.1.92) => Ext 8001 *FS dialplan =>* <extension name="public"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/external/$1(a)192.168.1.92"/> <action application="hangup"/> </condition> </extension> Added DID Provider IP and Asterisk IP in ACL. When I checked on Asterisk, got an error in sngrep, "401 Unauthorized". Attached image is of error. *On Asterisk =>* SIP.conf [freeswitch_1] type=peer host=192.168.1.70 port=5080 disallow=all allow=ulaw trustrpid=yes sendrpid=yes context=from-internal EXTENSIONS.conf exten => _XXX.,1,Dial(SIP/8001,30,oT) exten => _XXX.,n,Hangup Tried on port 5080 and 5060. Error both time same. My point is, this is Asterisk issue or Freeswitch? Please advice. -- Thanks & Regard Uday. Mobile:- +91-9377579349

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Call barge should allow the bargee to mute/unmute the caller (user B).
by Rajnikant Vanza 02 Nov '17

02 Nov '17

Hi all, I must fulfill the following requirements for call barge with mute/unmute features: As a user, I need the ability to "barge" in on a call already underway (between user 1 & user 2). When I "barge" in on the call, I want one user (say user 1) to hear me but not for the other (user 2) to hear me or what user 1 might say back to me (as if user 1 is muted). For e.g. Let's say two parties B and C are on an active call, and A is the party who initiates intercom/barge to party B. Now party B wants to talk to A and keep the party C on mute. We don't have such functionality either using intercom or eavesdrop in having the luxury for party B to mute/unmute C. The call barge functionality is working fine but I need to bargee (user B) able to mute/unmute user C. Can anybody please advise, how to achieve this functionality in FS? Thanks in advance, Best Regards, Rajnikant Vanza

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Masking conference recording
by Oleg Stolyar 01 Nov '17

01 Nov '17

Hi guys, In https://freeswitch.org/jira/browse/FS-5269 a command was added to mask session recordings. (uuid_record mask <path> and uuid_record unmask <path>) Is there a way to do the same with conference recording?

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How to call B-leg without opus rtpmap ?
by Zheka Polivoda 01 Nov '17

01 Nov '17

Hi all I'm using freeswitch to call from browser(webrtc) to some sip provider. So, the A-keg is webrtc opus codec and then I bridge it with the sip gateway. Sip gateway supports g711 only. And it chooses wrong codec and DTMF is not working between them. When bridging to sip, freeswitch sends such SDP 72a02659-6c1b-424e-b76a-0a9b16aecc2a Local SDP: 72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0 72a02659-6c1b-424e-b76a-0a9b16aecc2a o=FreeSWITCH 1509346805 1509346806 IN IP4 x.x.x.x 72a02659-6c1b-424e-b76a-0a9b16aecc2a s=FreeSWITCH 72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 x.x.x.x 72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0 72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 31568 RTP/AVP 102 9 0 8 103 101 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:102 opus/48000/2 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:102 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; sprop-maxcapturerate=8000; ptime=20; minptime=10; maxptim e=40; stereo=1 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:9 G722/8000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:0 PCMU/8000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:101 telephone-event/8000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:101 0-16 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=sendrecv Is it possible to send SDP without opus rtpmap? Because SIP gateway choose wrong telephone-event. the answer of sip gateway is 72a02659-6c1b-424e-b76a-0a9b16aecc2a 2017-10-30 15:46:13.248384 [DEBUG] sofia.c:7058 Remote SDP: 72a02659-6c1b-424e-b76a-0a9b16aecc2a v=0 72a02659-6c1b-424e-b76a-0a9b16aecc2a o=lvp 8000 8000 IN IP4 y.y.y.y 72a02659-6c1b-424e-b76a-0a9b16aecc2a s=SIP Call 72a02659-6c1b-424e-b76a-0a9b16aecc2a c=IN IP4 y.y.y.y 72a02659-6c1b-424e-b76a-0a9b16aecc2a t=0 0 72a02659-6c1b-424e-b76a-0a9b16aecc2a m=audio 22786 RTP/AVP 8 103 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:8 PCMA/8000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=rtpmap:103 telephone-event/48000 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=fmtp:103 0-16 72a02659-6c1b-424e-b76a-0a9b16aecc2a a=ptime:20 I guess there must be a=rtpmap:101 telephone-event/8000 Thanks

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Freeswitch video call recording.
by Nikhil 01 Nov '17

01 Nov '17

Hi I am trying to record video call with freeswitch. I am presently using freeswitch v 1.6.19. and am using mod_fsv. However my video recording doesnt work. I have been reading freeswitch confluence here - https://freeswitch.org/confluence/display/FREESWITCH/Video-recording and it seems that I need to checkout video-media-bug and built it. However I would like to know if this change has been merged and if so which version of freeswitch I need to use ? Cheers -./Nikhil -*-*-*-*-*-*-*- I learn a lot from my mistakes... and I keep learning new things everyday!

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