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Last dtmf digit
by Alex Lake 25 Jun '15

25 Jun '15

It's been a while since I got my hands dirty, but I'm struggling to remember how I can ask FS what the last dtmf digit detected was. I'm pretty certain it can be done, though! Can anyone please remind me? ;-)

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Update To and From Header of the legs after an uuid_bridge
by Alexander Haugg 25 Jun '15

25 Jun '15

Hi All, What can i do to change (fix) the from- and the to- header on SIP side after an uuid_bridge (SIP UPDATE or re-INVITE)? Thanks a lot!

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ICANN | privacy WHOIS
by EL 25 Jun '15

25 Jun '15

Already noticed the proposed ICANN policy changement? The new proposed rules: gnso.icann.org/en/issues/raa/ppsai-initial-05may15-en.pdf Check out: https://www.respectourprivacy.com/ -- EL

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how to transcode from g.729 to g.711a ?
by Jeff Chua 25 Jun '15

25 Jun '15

I'm attempting to call from a SIP phone (g.729) to the SIP gateway that connects g.711a to the PSTN. It's resulting in "SERVICE UNAVAILABLE" .... If I switch my SIP phone to use g.711a, everything works fine. Any sample on how to transcode 9.729 from my SIP phone to the freeswitch server that would then transcode from g.729 to g.711a to connect to the SIP gateway? I've tried g.729 from my SIP phone to call 1000 and 5000 on the freeswitch server and that worked. And I've bought license for g.729. Thanks, Jeff. ------------------------------------------------------------------------ INVITE sip:+6568465283@155.161.200.188 SIP/2.0 Via: SIP/2.0/TCP 155.161.11.194:54927;rport;branch=z9hG4bKPjDNkdtN2t3.xjdgQ-78gpmFRuBhnjA9Qd;alias Max-Forwards: 70 From: "1001" <sip:1001@155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y To: <sip:+6568465283@155.161.200.188> Contact: <sip:1001@155.161.11.194:54927;transport=TCP;ob> Call-ID: kQZCHnkGqshj5lmFItexEB8zNG1MA4W6 CSeq: 21273 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, norefersub User-Agent: Bria iOS 3.3.4 Content-Type: application/sdp Content-Length: 234 v=0 o=- 3643948404 3643948404 IN IP4 155.161.11.194 s=cpc_med c=IN IP4 155.161.11.194 t=0 0 m=audio 4004 RTP/AVP 18 101 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 ------------------------------------------------------------------------ INVITE sip:68465283@155.161.200.188:5070 SIP/2.0 Via: SIP/2.0/UDP 155.161.200.188:5080;rport;branch=z9hG4bKKX98pj6UB2crp Max-Forwards: 69 From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070> Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:gw+outbound1@155.161.200.188:5080;transport=udp;gw=outbound1> User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 231 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1001" <sip:1001@155.161.200.188:5070>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1434929593 1434929594 IN IP4 155.161.200.188 s=FreeSWITCH c=IN IP4 155.161.200.188 t=0 0 m=audio 30028 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Content-Length: 0 ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp From: "Extension 1001" <sip:FreeSWITCH@155.161.200.188:5070>;tag=KKS78e6eNpUpN To: <sip:68465283@155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324 Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33 CSeq: 77143746 INVITE Contact: <sip:68465283@155.161.200.188:5070> Retry-After: 1 Content-Length: 0 ------------------------------------------------------------------------ My config ... # vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/> # dialplan/default.xml <extension name="oubout_pstn"> <condition field="destination_number" expression="^\+65(.+)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/outbound1/$1"/> <action application="set" data="proxy_media=false"/> <action application="set" data="bypass_media=true"/> </condition> </extension> # sip_profiles/external.xml <gateways> <X-PRE-PROCESS cmd="include" data="external/*.xml"/> </gateways> <settings> <param name="disable-transcoding" value="false"/> ... </settings> # sip_profiless/external/example.xml <gateway name="outbound1"> <param name="realm" value="155.161.200.188:5070"/> <param name="register" value="false"/> <variables> <variable name="absolute_codec_string" value="PCMA" direction="outbound"/> </variables> <param name="reg-id" value="1"/> </gateway>

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Inbound calls one way audio remote handsets
by Jackson Fisher - Forward.IT 25 Jun '15

25 Jun '15

Hi All, I am after some assistance troubleshooting incoming one way audio with remote extensions. I have a remote office with the handsets registering over the internet back to the pbx Outgoing calls over SIP work without issue. I have the correct ports forwarded for SIP and RTP, however I am experiencing one way audio on the calls. Where should I start troubleshooting the issue? Is STUN required? Any help would be much appreciated. Thanks, Jackson Fisher [Email-signature.jpg] Jackson Fisher | Network Engineer A : Level 1 - Unit 8, 55-57 Lathlain Street, Belconnen ACT 2617 P : (02) 6162 0070 E : Jackson(a)forwardit.com.au<mailto:Jackson@forwardit.com.au> W : http://www.forwardit.com.au<http://www.forwardit.com.au/>

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VLC: Failed to change zoom Freeswitch 1.6
by Murugan Pandian 25 Jun '15

25 Jun '15

Hi, I try to play MP4 file using mod_vlc ,I am getting following error VLC: Failed to change zoom VLC: Failed to set on top VLC: Failed to change source AR VLC: Failed to change zoom VLC: Failed to change source AR But i can able to hear Audio only video i can't Full log 2015-06-24 16:06:29.808637 [NOTICE] mod_vlc.c:700 VLC Path is http http://www.html5rocks.com/en/tutorials/video/basics/devstories.mp4 Fontconfig warning: ignoring UTF-8: not a valid region tag 2015-06-24 16:06:30.728647 [INFO] mod_vlc.c:480 chroma: I420, width: 640, height: 386, pitches: 0, lines: 0 2015-06-24 16:06:30.728647 [INFO] mod_vlc.c:514 chroma: YUYV, width: 640, height: 386, pitches: 1280, lines: 386 VLC: Failed to change zoom VLC: Failed to set on top VLC: Failed to change source AR VLC: Failed to change zoom VLC: Failed to change source AR 2015-06-24 16:07:41.358701 [INFO] mod_vlc.c:524 cleanup 2015-06-24 16:07:41.358701 [NOTICE] switch_core_state_machine.c:315 sofia/internal/1005(a)x.x.x.x has executed the last dialplan instruction, hanging up. 2015-06-24 16:07:41.358701 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/internal/1005(a)x.x.x.x [CS_EXECUTE] [NORMAL_CLEARING] 2015-06-24 16:07:41.358701 [NOTICE] switch_core_session.c:1657 Session 1 (sofia/internal/1005(a)x.x.x.x) Ended 2015-06-24 16:07:41.358701 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/1005(a)x.x.x.x[CS_DESTROY]

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how to Play MP4 video file?
by Murugan Pandian 24 Jun '15

24 Jun '15

HI, How to play mp4 video file in freeswitch using dialplan or if any way to play video file in freeswitch

2 1

Websocket Load Balancing freeswitch
by Murugan Pandian 24 Jun '15

24 Jun '15

Hi, How i can handle load balancing websocket request and how to scale FS for WEBRTC

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Join us for ClueCon Weekly! Today (Wed Jun 24) at 1PM EST
by Ken Rice 24 Jun '15

24 Jun '15

Greetings FreeSWITCHers! Don¹t forget to join us for ClueCon Weekly today at 1PM EST (10AM PST)! This week¹s guest will be the 2600Hz Crew! Join us Live with your WebCam at https://conference.freeswitch.org/verto/ and call 888! Or watch it via YouTube at: https://youtu.be/VC44I_FdLdk Hang Around after the live broadcast to help us test! Also, Don¹t Forget ClueCon! If you havent registered yet go to https://cluecon.com and register today! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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calling again if the receiver hangsup
by jam li 24 Jun '15

24 Jun '15

Hi, I am using freeswitch server for android softphone development. When i am calling from my softphone to a mobile number, and if the receiver of the call hangs up without answering. then i am getting voice mail message from the receiver end. and The mobile number at the receiver end is receiving the call again. please guide me which channel variable i must set in my sip providers dialplan, so that the call is established only once. This behavior is not happening when i call between freeswitch extention to freeswitch extention. Regards.

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