May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Silence Detection in Recording
by Jeevan Singh Dhami 06 Jun '16

06 Jun '16

Hi, I have recorded the voice using freeswitch and their found silence in recording if recording goes below *given silence threshold. after that freeswitch stop the recording .but there was no indication given by freeswitch for silence detection .Could you please tell me there is any way to find the indication parameters in freeswitch. Either any solution for knowing silence found in recording in freeswitch.* *please share the solution in same mail .i will appreciate your feedback. * -- Thanks & Regards Jeevan Singh Dhami Software Engineer – Development

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Bridge Call and Join Conference
by Colin Morelli 04 Jun '16

04 Jun '16

So, I'm trying to setup certain calls such that they enter into a conference asap. I have the process working using bridging conferences right now, which is okay, but bridging conferences forces the dialing leg of the call to go to an answered state immediately, while the bridge leg is still ringing. So, while everything seems to work correctly, it messes with some of my reconciliation/reporting process with another database that are fed off of channel state events. Essentially it looks like every call was answered. So I was curious to know if there's a way to make a call that creates a bridged leg, and then as soon as the leg is answered immediately joins both legs into a conference. Is the only way to do it to listen for the channel_bridge events and then transfer the call myself? Best, Colin

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Low cost voip phone ?
by Shawn Wheeler 04 Jun '16

04 Jun '16

I am looking for recommendations for a low cost voip handset. I have been using grandstream but can't seem to get past the delay when making a call. Thanks in advance Sent from my Verizon, Samsung Galaxy smartphone

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Sharing presence between FS boxes
by Chris B. Ware 04 Jun '16

04 Jun '16

Hi, On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358). If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B? By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_" as I read on wiki. Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared between boxes, and even presence works sometimes, but is not stable. Here my internal sip profiles config: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="configuration" description="Various Configuration"> <configuration name="sofia.conf" description="Sofia Endpoint"> <profiles> <profile name="internal"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="$${recordings_dir}"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="$${internal_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${internal_tls_port}"/> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/> <param name="challenge-realm" value="auto_from"/> <param name="outbound-proxy" value="sip::5060"/> </settings> </profile> </profiles> </configuration> </section> </document> Can somebody help?

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Re: [Freeswitch-users] Presence with shared database
by Adam Kuśmirek 04 Jun '16

04 Jun '16

Hello, Yes we enabled manage-presence. According to documentation dbname is used to share presence between profiles. What we want to achieve is to share presence between hosts. We investigated some presence source code and found that there are sql queries that ask for specific FS host. We modified SELECT statements and UPDATES of version parameter, so they have no host in WHERE clause. After mod_sofia recompilation everything works fine. It seems that the only way to share presence between hosts is to set the same hostname in switch.xml for every FS node, but we cannot do this because we need host information for other services. I wonder why FS checks whether the subscription was made on a host that sets up the connection. Maybe this is a bug in the concept and it should be removed. Regards Adam Kusmirek ---------- Wiadomość przekazana dalej ---------- >> > From: "Ítalo Rossi" <italorossib(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 7 Aug 2015 22:42:37 -0500 > Subject: Re: [Freeswitch-users] Presence with shared database > Hard to say without more info, where's your configs? > > Did you enabled the manage-presence in your profile? and set the dbname? > > >

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Ringback after pre_answer without bridge?
by Dmitry Sytchev 03 Jun '16

03 Jun '16

Hi all! I'd like to resolve some unclear ringback behavior in early media mode. I'm trying to implement ringback while no bridge active. A calls FS, FS does something, then searching for B and bridges A to B. I'd like to start playing custom ringback immediately after call from A gets into FS, not when we start bridge to B. I'm doing something like that: <action application="set" data="ringback=%(2000, 4000, 440.0, 480.0)"/> <action application="set" data="instant_ringback=true"/> <action application="set" data="bridge_early_media=true"/> <action application="pre_answer"/> <action application="sleep" data="10000"/> ... <action application="bridge" data="some bridge data"/> There is no ringback to A in such scenario until bridge is called in further dialplan processing. I've tried all combinations of ignore/bridge early media, but with no luck. Pre_answer can be called directly or by FS when call is recorded, with no difference. There is a wiki page that recommends usage of ring_ready: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+ring_re… ring_ready works, but it can give only Ringing to calling side, so I can't pass custom RBT to calling party. It would be enough for me, but once somebody calls pre_answer, no early media is passed until bridge attempt. What can be done to resolve this? Thanks in advance. -- Best regards, Dmitry Sytchev, IT Engineer

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Windows build
by Gregor Nanger 03 Jun '16

03 Jun '16

Hi! I want to build Freeswitch on windows with visual studio 2015. Where should I start if I get 600 errors when try to Rebuild All. I opened solution and start Rebuild All, but I get so many errors that I belive that I am doing something wrong. Mainl yre errors regarding: Cannot open source file.... Best regards, Gregor

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SIP Message question
by Nagorny, Dimitry 01 Jun '16

01 Jun '16

Hello List, AFAIK, all INVITE SIP messages (INVITE sip:100@mydomain.com) are going through the set up dialplans. Are all MESSAGE SIP messages (MESSAGE sip:100@mydomain.com) also go through these dialplans? If no, how can I handle them? This might be a stupid question but any help is still appreciated. Best Regards Dimitry Nagorny Trainee

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ClueCon Speaker Registration Deadline 2 Weeks Away!!!!
by Anthony Minessale 31 May '16

31 May '16

Don't forget to get your proposal in. E-mail speakers(a)cluecon.com -- 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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SDP with double m=audio lines and with the same port
by Ján Füri 31 May '16

31 May '16

Hello, My freeswitch's INVITE sdp offers two m=audio lines. (RTP/SAVP and RTP/AVP). That's OK, because I want it so ;) But both use the same audio port number. ** Is that possible to change this behavior ? To use different audio port numbers for each m=audio line ... Please check my example below. In SDP from my freeswitch, both m-lines have *m=audio **22284*. The reason, why I want to change this is WebRTC and RTPengine. RTPengine changes this INVITEs to RTP/SAVPF. And the changed RTP/SAVPF is then not acceptable for chrome browsers. Firefox works well (so far). I already reported this to the RTPengine but they say the problem is with INVITE from freeswitch because the m-audio lines use the same port. My example : incoming calls -> Media Server (freeswitch) -> Kamailio (sip and websocket proxy) -> WebRTP and SIP clients I have found a feature *media_webrtc=true* so I could avoid using rtpengine for webrtc clients (that would be really awesome), but then classic sip clients are offered only with RTP/SAVPF and most sip phones do not know RTP/SAVPF so the calls are rejected :( INVITE SDP from freeswitch: v=0. o=SBC 1464589775 1464589776 IN IP4 <freeswitch-public-ip>. s=SBC. c=IN IP4 <freeswitch-public-ip>. t=0 0. *m=audio 22284 RTP/SAVP 8 0 9 3 18.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtcp:22285 IN IP4 <freeswitch-public-ip>. a=crypto:1 AEAD_AES_256_GCM_8 inline:Q/O+iUzZ9AZLcrADA7w/XcpkmuW6yArr1vsaWc0coTyWUIRLX2qfCA7XDAs. a=crypto:2 AEAD_AES_128_GCM_8 inline:mBb/SmBuSQNSm8lC5giUfZMCv0ZYINvsQiX2sw. a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:ahOTNSsdmHLIOBqvUyGyNjd+gDfAE/+jA6w7XwzGWxBMjtzf5akXNvM/OGy0jQ. a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:k7002pXV/SUT7JHhZTYaMV5keUTp1EP57M4rcNZnAoZZzsceTXA. a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:bHcV4E/OnzMkNeaFplPWt4RELILYZeGlifnJNlRV. a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:zJkowU1tc5rQR5BPpg2m3eE97ZqXLFFJc1Agh89XuZynPFMrXhO266+eMZCd2A. a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:2DwGI3UAPsLFBS5sJBsc+pzEsITQwDHCvB0u7pK/XEE3G8swsrw. a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:Jen6Gm56Z+4RTRmfpXTpSpoG5bduyythl1j21a15. a=crypto:9 AES_CM_128_NULL_AUTH inline:Nn83liGQY7eY8LGs9qz/7EoPiHOpRgwY8H7Ts/b5. a=ptime:20. *m=audio 22284 RTP/AVP 8 0 9 3 18.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtcp:22285 IN IP4 <freeswitch-public-ip>. a=ptime:20. INVITE SDP from RTPengine : v=0 o=SBC 1464589775 1464589776 IN IP4 212.232.17.66 s=SBC c=IN IP4 <rtpengine-public-ip> t=0 0 *m=audio 31836 RTP/SAVPF 8 0 9 3 18* a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:31837 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F a=ice-ufrag:NauDcjVU a=ice-pwd:AiHk6LF4tE0GElyFMWtLeod1wH a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 31836 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 31837 typ host *m=audio 31866 RTP/SAVPF 8 0 9 3 18* a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:31867 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F a=ice-ufrag:YxCyPhQV a=ice-pwd:2vYt8YaMhIp2DSLPeYOKDqqBX0 a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 31866 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 31867 typ host

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