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Ring issue
by Uday kumar 04 Oct '16

04 Oct '16

Hi, I have installed FS 1.6 on server. I registered two extensions from two different network. My server is on public IP. When I call calling on extension between each other, in FS log its showing ringing but at caller side its silent and also no ring at other end. Pleased advice. I did some changes sofia:- ext-rtp-ip=auto-nat ext-sip-ip=auto-nat to ext-rtp-ip=Public-IP ext-sip-ip=Public-IP Below is my of sofia status profile internal and sofia status profile external [root@192 sip_profiles]# fs_cli -rx "sofia status profile internal" ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts Public-IP,Public-IP Dialplan XML Context public Challenge Realm auto_from RTP-IP Public-IP SIP-IP Public-IP URL sip:mod_sofia@Public-IP:5060 BIND-URL sip:mod_sofia@Public-IP:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@Public-IP:5066;transport=ws WSS-BIND-URL sips:mod_sofia@Public-IP:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT OPUS,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT true CALLS-IN 2 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 1 REGISTRATIONS 7 [root@192 sip_profiles]# fs_cli -rx "sofia status profile external" ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP Public-IP SIP-IP Public-IP URL sip:mod_sofia@Public-IP:5080 BIND-URL sip:mod_sofia@Public-IP:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN OPUS,G722,PCMU,PCMA,GSM CODECS OUT OPUS,G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 *I changed my server IP with Public-IP for security. * Thanks Uday. -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349

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Get disconnected message
by Antonio Silva 04 Oct '16

04 Oct '16

Hi all, I've a simple lua esl script to handle fs events outside the main program, were: while(con:connected() == 1) do -- get and act on con:events -- break the loop on event SERVER_DISCONNECTED or SHUTDOWN done It works quite well, but at some points the loop just breaks without the receiving any control event (disconnected or shutdown) and i can't figure out why it breaks. I got it running for 5 h... Is it possible to get the cause why con:connected() is returning 0? or some log i could enable? The scripts is running on debian 8.6 and Lua 5.2.3. Thanks, António

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can I reload switch.conf.xml dynamically
by Dave Horton 04 Oct '16

04 Oct '16

I want to change the value of the switchname param in switch.conf.xml. Do I need to restart freeswitch in order for that to take affect, or can I dynamically reload it so it takes affect? Dave

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mod_avmd with mod_com_g729
by Chad 04 Oct '16

04 Oct '16

Hello FreeSWITCH Users, Just to get this out of the way: FreeSWITCH 1.6.10 installed via CentOS 7 repo I’m curious to see if anyone is using the g729 codec with mod_avmd. I’m doing some testing for a project here which is going to require some answering machine detection. We would like to use the g729 codec to keep bandwidth down but I am running into an issue where avmd detects a false beep right after being started on the channel via the ESL. This happens about 50% of the time but I haven’t been able to reproduce the issue using PCMU. In the console I see avmd start, the media bug gets set and the g729 encoder / decoder gets allocated. Immediately after this happens a beep is detected causing my application to go in the wrong direction. Any input is welcome. Thanks! Chad

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How to pass variable to Loopback Channel with loopback_export?
by Peter Steinbach 04 Oct '16

04 Oct '16

Hello, I would like to do initiate a call the following way: {variable1=xxx,variable2=yyy}sofia/gateway/number &bridge(loopback/number2) Variables are not passed to the loopback channel, as expected. So I tried the loopback_export the following way. {variable1=xxx,variable2=yyy,loopback_export=variable1}sofia/gateway/number &bridge(loopback/number2) However, variables are not passed to the loopback channel. Anybody knows how to pass variables to the loopback channel? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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FreeSwitch & Simulcast
by Mirko Brankovic 04 Oct '16

04 Oct '16

Hi, I'm wondering where do FS stands now and what is the plan for future? I saw there were PlanB <https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00> and Unified Plan <https://tools.ietf.org/html/draft-roach-mmusic-unified-plan-00> but I guess none passed the point of proposal I guess. Now I see some different "webrtc standards" as *rid* in sdp <https://webrtcstandards.info/simulcast-and-sending-multiple-video-resolutio…> . Simulcast is the future I guess, like it or not i guess. My general question is will FS go in SFU direction and why not ??? :D -- Regards, Mirko ¯\_(ツ)_/¯

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Sofia Status Question
by Shawn Wheeler 03 Oct '16

03 Oct '16

I have a few machines running FS. The last one I brought online has additional line pointing to 127.0.0.1 when I run sofia status on the CLI. Below I have the same command from a different machine. My question is the 127.x.x.x. line hurting anything? I am not sure how it got there and less sure of how to make it go away. Thanks in advance for the help. Shawn freeswitch@internal> sofia status Name Type Data State ================================================================================================= 127.0.0.1 alias internal ALIASED external profile sip:mod_sofia@192.168.51.195:5080 RUNNING (0) external::8016664500 gateway sip:8016864500@west.broadcore.com REGED external::8016664501 gateway sip:8016864501@west.broadcore.com REGED internal profile sip:mod_sofia@10.171.72.3:5060 RUNNING (0) ================================================================================================= 2 profiles 1 alias freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@192.168.51.55:5080 RUNNING (0) external::8018268048 gateway sip:8018268348@west.broadcore.com REGED external::8018268047 gateway sip:8018268347@west.broadcore.com REGED 192.168.40.55 alias internal ALIASED internal profile sip:mod_sofia@10.111.72.3:5060 RUNNING (0) ================================================================================================= 2 profiles 1 alias

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Audio Control on mp3/wav
by Jai Rangi 03 Oct '16

03 Oct '16

Checking to see if it is possible to have controls on audio files played with playback or filestream. Expected controls are keys to rewind, forward, fast-forward, pause, play. Jai

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issue with ilbc and ptime between a call between webrtc client and freeswitch
by Caragea Silviu 03 Oct '16

03 Oct '16

Hello, I have a call coming from FS and going via sip to a webrtc compliant client. Works fine as time the codec is not ILBC. If codec is ILBC is failing because Webrtc consider the codec as being invalid because of the ptime line. Inside their code: bool IsILBCRateValid(int rate, int frame_size_samples) { if (((frame_size_samples == 240) || (frame_size_samples == 480)) && (rate == 13300)) { return true; } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) && (rate == 15200)) { return true; } else { return false; } } This is failing because rate will be 13300 and frame_size_samples will be 160. frame_size_samples = (8000 / 1000) * 20 which is 160. How I can make fs to advertise correctly that the codec ILBC is 30 ms one and not 20 ? Here is the sdp that's generated: v=0 o=FreeSWITCH 1475470534 1475470535 IN IP4 91.231.138.218 s=FreeSWITCH c=IN IP4 91.XXX.XXX.XXX t=0 0 a=msid-semantic: WMS 5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a=end-of-candidates m=audio 16430 RTP/SAVPF 102 0 103 101 104 a=rtpmap:102 iLBC/8000 a=fmtp:102 mode=30 a=rtpmap:0 PCMU/8000 a=rtpmap:103 isac/16000 a=fmtp:103 ibitrate=32000;maxbitrate=53400 a=rtpmap:101 telephone-event/8000 a=rtpmap:104 telephone-event/16000 a=fingerprint:sha-256 B9:24:01:90:AD:E6:43:60:10:F9:CA:3A:85:86:26:91:A6:41:A8:77:3B:F0:39:20:3C:50:0F:A3:5E:1F:48:03 a=setup:actpass a=rtcp-mux a=rtcp:16430 IN IP4 91.xxx.xxxx.xxx a=ssrc:4159978388 cname:tTLcMVJxIsF1gi8j a=ssrc:4159978388 msid:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a0 a=ssrc:4159978388 mslabel:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a=ssrc:4159978388 label:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0va0 a=ice-ufrag:VWQeyw6k2xxshnNB a=ice-pwd:6oDJad6JwBACKzMvlsAKOwUH a=candidate:8788212768 1 udp 659136 91.XXX.XXX.XXX16430 typ host generation 0 a=candidate:8788212768 2 udp 659136 91.XXX.XXX.XXX16430 typ host generation 0 a=silenceSupp:off - - - - a=ptime:20 Silviu

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Re: [Freeswitch-users] *.tmp.xml.fsxml Files
by Luke Milbourne 03 Oct '16

03 Oct '16

Hi All, Below is an example of some of the files... da12b04e-a164-4fca-a1b3-9e610d98f826.tmp.xml.fsxml e0cf7925-bbc6-4de6-8cb2-0bc58b3b9854.tmp.xml.fsxml e6261ce6-42ac-4b85-8fc5-f31520073caa.tmp.xml.fsxml e9292385-1580-4a52-acaf-76fb1d7bd55d.tmp.xml.fsxml ebadf784-f39a-4478-b2a6-84ca5afc9e29.tmp.xml.fsxml ec43990e-4e41-4c0f-bd7f-906f18be94e0.tmp.xml.fsxml ee20d167-5cd2-46cd-b564-e9d448ec2ac8.tmp.xml.fsxml ee387849-14af-4531-bb25-171a807a6ad0.tmp.xml.fsxml f13071b6-6da4-4f09-94e2-3530f153f9f0.tmp.xml.fsxml f1cd4cb0-ec58-4035-8583-b42ccc40fbbc.tmp.xml.fsxml Kind regards, Luke On 3 October 2016 at 17:14, <freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: > > *.tmp.xml.fsxml Files (Michael Jerris) > 2. Re: issue with ilbc and ptime between a call between webrtc > client and freeswitch (Michael Jerris) > 3. Re: ?*.tmp.xml.fsxml Files (Michael Jerris) > 4. Re: FreeSwitch & Simulcast (Michael Jerris) > 5. Re: issue with ilbc and ptime between a call between webrtc > client and freeswitch (Brian West) > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 11:58:51 -0400 > Subject: Re: [Freeswitch-users] *.tmp.xml.fsxml Files > we never make files with that pattern, we will make files with .fsxml.tmp > there in some cases. What are the exact filenames? > > On Oct 3, 2016, at 4:37 AM, Luke Milbourne <luke.milbourne(a)gmail.com> > wrote: > > Hi All, > > Can anyone confirm what the purpose of these files are in FreeSWITCH's log > directory? > > They're all empty so no clues in the actual files. > > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 12:02:06 -0400 > Subject: Re: [Freeswitch-users] issue with ilbc and ptime between a call > between webrtc client and freeswitch > libc@20 is a totally valid rate for libc. If they are rejecting that, > you should file a bug with them. Depending on how you are negotiating > codecs you could force it by our normal codec negotiation settings. > > > > On Oct 3, 2016, at 5:40 AM, Caragea Silviu <silviu.cpp(a)gmail.com> wrote: > > > > Hello, > > > > I have a call coming from FS and going via sip to a webrtc compliant > client. > > Works fine as time the codec is not ILBC. If codec is ILBC is failing > because > > Webrtc consider the codec as being invalid because of the ptime line. > > > > Inside their code: > > > > bool IsILBCRateValid(int rate, int frame_size_samples) { > > if (((frame_size_samples == 240) || (frame_size_samples == 480)) && > > (rate == 13300)) { > > return true; > > } else if (((frame_size_samples == 160) || (frame_size_samples == > 320)) && > > (rate == 15200)) { > > return true; > > } else { > > return false; > > } > > } > > > > This is failing because rate will be 13300 and frame_size_samples will > be 160. > > > > frame_size_samples = (8000 / 1000) * 20 which is 160. > > > > How I can make fs to advertise correctly that the codec ILBC is 30 ms > one and not 20 ? > > > > Here is the sdp that's generated: > > > > v=0 > > o=FreeSWITCH 1475470534 1475470535 IN IP4 91.231.138.218 > > s=FreeSWITCH > > c=IN IP4 91.XXX.XXX.XXX > > t=0 0 > > a=msid-semantic: WMS 5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v > > a=end-of-candidates > > m=audio 16430 RTP/SAVPF 102 0 103 101 104 > > a=rtpmap:102 iLBC/8000 > > a=fmtp:102 mode=30 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:103 isac/16000 > > a=fmtp:103 ibitrate=32000;maxbitrate=53400 > > a=rtpmap:101 telephone-event/8000 > > a=rtpmap:104 telephone-event/16000 > > a=fingerprint:sha-256 B9:24:01:90:AD:E6:43:60:10:F9: > CA:3A:85:86:26:91:A6:41:A8:77:3B:F0:39:20:3C:50:0F:A3:5E:1F:48:03 > > a=setup:actpass > > a=rtcp-mux > > a=rtcp:16430 IN IP4 91.xxx.xxxx.xxx > > a=ssrc:4159978388 cname:tTLcMVJxIsF1gi8j > > a=ssrc:4159978388 msid:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a0 > > a=ssrc:4159978388 mslabel:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v > > a=ssrc:4159978388 label:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0va0 > > a=ice-ufrag:VWQeyw6k2xxshnNB > > a=ice-pwd:6oDJad6JwBACKzMvlsAKOwUH > > a=candidate:8788212768 1 udp 659136 91.XXX.XXX.XXX16430 typ host > generation 0 > > a=candidate:8788212768 2 udp 659136 91.XXX.XXX.XXX16430 typ host > generation 0 > > a=silenceSupp:off - - - - > > a=ptime:20 > > > > Silviu > > ____________________________________________________________ > _____________ > > 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 > > > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 12:04:46 -0400 > Subject: Re: [Freeswitch-users] *.tmp.xml.fsxml Files > are you sure thats the file pattern? we do put uuid.tmp.xml files in > there in some cases.. but not uuid.tmp.xml.fsxml. > > > On Oct 3, 2016, at 7:01 AM, Luke Milbourne <luke.milbourne(a)gmail.com> > wrote: > > Hi Josh, > > Hope your keeping well mate. > > I know about that one :) > > These ones are suffixed ".tmp.xml.fsxml" with what looks like a UUID as > there filename. > > But they're all empty....just wondered if anyone else knows what they're > for. > > Thanks, > Luke > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 12:07:34 -0400 > Subject: Re: [Freeswitch-users] FreeSwitch & Simulcast > Currently there are lots of good tools to do SFU, I’m not sure what the > upside is to us doing SFU as well that is not already covered by other > tools. Its a similar discussion to people who want FS to be a proxy. If > there is sufficient demand and funding to do the work, we would consider > it. Lacking the funding to do this work I don’t see the drive to do it. > > > On Oct 3, 2016, at 10:45 AM, Mirko Brankovic <mirkobrankovic(a)gmail.com> > wrote: > > Hi, > I'm wondering where do FS stands now and what is the plan for future? > > I saw there were PlanB > <https://tools.ietf.org/html/draft-uberti-rtcweb-plan-00> and Unified Plan > <https://tools.ietf.org/html/draft-roach-mmusic-unified-plan-00> but I > guess none passed the point of proposal I guess. > > Now I see some different "webrtc standards" as *rid* in sdp > <https://webrtcstandards.info/simulcast-and-sending-multiple-video-resolutio…> > . > > Simulcast is the future I guess, like it or not i guess. > > My general question is will FS go in SFU direction and why not ??? :D > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 3 Oct 2016 11:14:16 -0500 > Subject: Re: [Freeswitch-users] issue with ilbc and ptime between a call > between webrtc client and freeswitch > its ilbc@20i and it should set mode=20 > > /b > > > On Mon, Oct 3, 2016 at 11:02 AM, Michael Jerris <mike(a)jerris.com> wrote: > >> libc@20 is a totally valid rate for libc. If they are rejecting that, >> you should file a bug with them. Depending on how you are negotiating >> codecs you could force it by our normal codec negotiation settings. >> >> >> > On Oct 3, 2016, at 5:40 AM, Caragea Silviu <silviu.cpp(a)gmail.com> >> wrote: >> > >> > Hello, >> > >> > I have a call coming from FS and going via sip to a webrtc compliant >> client. >> > Works fine as time the codec is not ILBC. If codec is ILBC is failing >> because >> > Webrtc consider the codec as being invalid because of the ptime line. >> > >> > Inside their code: >> > >> > bool IsILBCRateValid(int rate, int frame_size_samples) { >> > if (((frame_size_samples == 240) || (frame_size_samples == 480)) && >> > (rate == 13300)) { >> > return true; >> > } else if (((frame_size_samples == 160) || (frame_size_samples == >> 320)) && >> > (rate == 15200)) { >> > return true; >> > } else { >> > return false; >> > } >> > } >> > >> > This is failing because rate will be 13300 and frame_size_samples will >> be 160. >> > >> > frame_size_samples = (8000 / 1000) * 20 which is 160. >> > >> > How I can make fs to advertise correctly that the codec ILBC is 30 ms >> one and not 20 ? >> > >> > Here is the sdp that's generated: >> > >> > v=0 >> > o=FreeSWITCH 1475470534 1475470535 IN IP4 91.231.138.218 >> > s=FreeSWITCH >> > c=IN IP4 91.XXX.XXX.XXX >> > t=0 0 >> > a=msid-semantic: WMS 5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v >> > a=end-of-candidates >> > m=audio 16430 RTP/SAVPF 102 0 103 101 104 >> > a=rtpmap:102 iLBC/8000 >> > a=fmtp:102 mode=30 >> > a=rtpmap:0 PCMU/8000 >> > a=rtpmap:103 isac/16000 >> > a=fmtp:103 ibitrate=32000;maxbitrate=53400 >> > a=rtpmap:101 telephone-event/8000 >> > a=rtpmap:104 telephone-event/16000 >> > a=fingerprint:sha-256 B9:24:01:90:AD:E6:43:60:10:F9: >> CA:3A:85:86:26:91:A6:41:A8:77:3B:F0:39:20:3C:50:0F:A3:5E:1F:48:03 >> > a=setup:actpass >> > a=rtcp-mux >> > a=rtcp:16430 IN IP4 91.xxx.xxxx.xxx >> > a=ssrc:4159978388 cname:tTLcMVJxIsF1gi8j >> > a=ssrc:4159978388 msid:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v a0 >> > a=ssrc:4159978388 mslabel:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0v >> > a=ssrc:4159978388 label:5K3pT3kGUXTisXNhdMYGtMP1VGBmzT0va0 >> > a=ice-ufrag:VWQeyw6k2xxshnNB >> > a=ice-pwd:6oDJad6JwBACKzMvlsAKOwUH >> > a=candidate:8788212768 1 udp 659136 91.XXX.XXX.XXX16430 typ host >> generation 0 >> > a=candidate:8788212768 2 udp 659136 91.XXX.XXX.XXX16430 typ host >> generation 0 >> > a=silenceSupp:off - - - - >> > a=ptime:20 >> > >> > Silviu >> > ____________________________________________________________ >> _____________ >> > 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 >> >> >> _________________________________________________________________________ >> 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 >> > > > > -- > > *Brian West* > brian(a)freeswitch.org > > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) > http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) > https://www.gofundme.com/freeswitch_ubuntu > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest > > _______________________________________________ > 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 > > -- Luke Milbourne <luke.milbourne(a)gmail.com> Tel: 07857154817 Google Talk/Email: luke.milbourne(a)gmail.com

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