May 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] can't hear audio when dialing 9198 (Tetris audio test)
by Jeff Leung 05 May '14

05 May '14

By default you’ll also need 16384-32767 opened up too for the RTP ports. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Moishe Grunstein Sent: Monday, May 5, 2014 07:55 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] can't hear audio when dialing 9198 (Tetris audio test) There is more than 5060 that needs to be opened, see https://wiki.freeswitch.org/wiki/Firewall Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified <http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Steven Ayre Sent: Monday, May 05, 2014 10:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] can't hear audio when dialing 9198 (Tetris audio test) Also check the IP in the logged "remote SDP" message (which is before what you're posted). Verify that it's an IP actually reachable from the server. A common problem is having a client that's behind NAT sending the LAN IP which isn't reachable over the internet, rather than it's external IP. On 5 May 2014 14:55, Michael Jerris <mike(a)jerris.com> wrote: do you have udp ports open for rtp? On May 4, 2014, at 9:03 PM, Jeff Stark <jstark888(a)gmail.com> wrote: Hi, I installed FreeSwitch 1.2.23 on an Amazon EC2 instance (running Amazon Linux). I opened port 5060 for UDP to the server. I then registered as user 1005 using SipDroid on my phone via WiFi. At first when I dialed 9198, I could hear the Tetris music. I could also hear the music on hold audio (when I dialed 9664). However, the echo test failed. Therefore I tried changing the config for the server according to this: http://wiki.freeswitch.org/wiki/Amazon_EC2 However, after doing that not only did the echo test not work, but neither did the Tetris or music on hold tests. So I returned the config back to what it was originally. Unfortunately I still can't hear the Tetris or music on hold audio. I have tried using both LinPhone and XLite softphones as alternatives, but I can't hear the audio on those either. Signalling appears to work. I can dial an extension on a registered softphone and have another registered softphone ring, but there is no audio. I would appreciate any help or insight anyone has on what the problem might be. Also, I have tried using G722, PCMA, PCMU and GSM audio codecs, but none have worked. All the softphones I have tried are able to register without any noticeable problems. Thanks. Jeff Stark Below is part of the log for a typical session where I can't hear the Tetris audio. (I have masked the IP addresses with descriptive labels): 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 Dialplan: sofia/internal/1005@<FreeSwitch server Public IP> Regex (PASS) [tone_stream] destination_number(9198) =~ /^9198$/ break=on-false 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 Dialplan: sofia/internal/1005@<FreeSwitch server Public IP> Action answer() 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 Dialplan: sofia/internal/1005@<FreeSwitch server Public IP> Action playback({loops=10}tone_stream://path=${base_dir}/conf/tetris.ttml) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/1005@<FreeSwitch server Public IP>) State Change CS_ROUTING -> CS_EXECUTE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/1005@<FreeSwitch server Public IP>) State ROUTING going to sleep 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1005@<FreeSwitch server Public IP>) Running State Change CS_EXECUTE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1005@<FreeSwitch server Public IP>) State EXECUTE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] mod_sofia.c:243 sofia/internal/1005@<FreeSwitch server Public IP> SOFIA EXECUTE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1005@<FreeSwitch server Public IP> Standard EXECUTE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> hash(insert/<FreeSwitch server Private IP>-spymap/1005/84a7c5c8-d251-11e3-9565-7f053ccdf4d9) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> hash(insert/<FreeSwitch server Private IP>-last_dial/1005/9198) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> hash(insert/<FreeSwitch server Private IP>-last_dial/global/84a7c5c8-d251-11e3-9565-7f053ccdf4d9) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> export(RFC2822_DATE=Fri, 02 May 2014 23:28:51 +0000) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_channel.c:1242 EXPORT (export_vars) [RFC2822_DATE]=[Fri, 02 May 2014 23:28:51 +0000] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> answer() 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:5282 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:3190 Set Codec sofia/internal/1005@<FreeSwitch server Public IP> G722/8000 20 ms 160 samples 64000 bits 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_codec.c:111 sofia/internal/1005@<FreeSwitch server Public IP> Original read codec set to G722:9 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:5451 Set 2833 dtmf send/recv payload to 101 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:3449 AUDIO RTP [sofia/internal/1005@<FreeSwitch server Public IP>] <FreeSwitch server Private IP> port 23728 -> <Softphone Public IP> port 50062 codec: 9 ms: 20 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_rtp.c:2040 Starting timer [soft] 160 bytes per 20ms 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:3716 Set 2833 dtmf send payload to 101 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:3722 Set 2833 dtmf receive payload to 101 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia_glue.c:3749 sofia/internal/1005@<FreeSwitch server Public IP> Set rtp dtmf delay to 40 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [NOTICE] sofia_glue.c:4360 Pre-Answer sofia/internal/1005@<FreeSwitch server Public IP>! 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_channel.c:3368 (sofia/internal/1005@<FreeSwitch server Public IP>) Callstate Change RINGING -> EARLY 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] mod_sofia.c:866 Local SDP sofia/internal/1005@<FreeSwitch server Public IP>: 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 v=0 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 o=FreeSWITCH 1399049603 1399049604 IN IP4 <FreeSwitch server Private IP> 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 s=FreeSWITCH 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 c=IN IP4 <FreeSwitch server Private IP> 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 t=0 0 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 m=audio 23728 RTP/AVP 9 101 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 a=rtpmap:9 G722/8000 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 a=rtpmap:101 telephone-event/8000 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 a=fmtp:101 0-16 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 a=ptime:20 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 a=sendrecv 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] sofia.c:5815 Channel sofia/internal/1005@<FreeSwitch server Public IP> entering state [completed][200] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_core_session.c:871 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [NOTICE] mod_dptools.c:1225 Channel [sofia/internal/1005@<FreeSwitch server Public IP>] has been answered 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_channel.c:3647 (sofia/internal/1005@<FreeSwitch server Public IP>) Callstate Change EARLY -> ACTIVE 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 EXECUTE sofia/internal/1005@<FreeSwitch server Public IP> playback({loops=10}tone_stream://path=/usr/local/freeswitch/conf/tetris.ttml) 2014-05-02 23:28:51.111375 [DEBUG] switch_event.c:1661 Parsing variable [loops]=[10] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.111375 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@16000hz 1 channels 20ms 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.331419 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.331419 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.331419 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:28:51.331419 [DEBUG] sofia.c:5815 Channel sofia/internal/1005@<FreeSwitch server Public IP> entering state [ready][200] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.331345 [DEBUG] switch_core_session.c:1016 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [NOTICE] sofia.c:737 Hangup sofia/internal/1005@<FreeSwitch server Public IP> [CS_EXECUTE] [NORMAL_CLEARING] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [KILL] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_session.c:1351 Send signal sofia/internal/1005@<FreeSwitch server Public IP> [BREAK] 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_ivr_play_say.c:1717 done playing file {loops=10}tone_stream://path=/usr/local/freeswitch/conf/tetris.ttml 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_session.c:2810 sofia/internal/1005@<FreeSwitch server Public IP> skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1005@<FreeSwitch server Public IP>) State EXECUTE going to sleep 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1005@<FreeSwitch server Public IP>) Running State Change CS_HANGUP 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_state_machine.c:678 (sofia/internal/1005@<FreeSwitch server Public IP>) Callstate Change ACTIVE -> HANGUP 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_state_machine.c:680 (sofia/internal/1005@<FreeSwitch server Public IP>) State HANGUP 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] mod_sofia.c:506 Channel sofia/internal/1005@<FreeSwitch server Public IP> hanging up, cause: NORMAL_CLEARING 84a7c5c8-d251-11e3-9565-7f053ccdf4d9 2014-05-02 23:29:31.351377 [DEBUG] switch_core_state_machine.c:48 sofia/internal/1005@<FreeSwitch server Public IP> Standard HANGUP, cause: NORMAL_CLEARING _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.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

1 0

Freeswitch with Net2Phone
by Umair Khan 05 May '14

05 May '14

Hi All, Does someone used net2phone with freeswitch. I am having no audio problem, below is my sip trace and sofia setting. Please let me know did I miss something? SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.178.239.32;branch=z9hG4bKZ38e140yHZSye;received=208.86.251.68;rport=5060 From: "Prod" <sip:Prod@10.178.239.32>;tag=epH2avec6ZXmm To: <sip:99901134346556@66.33.147.149 >;tag=sbcsipuas_1_C50986_20140501113400982_b12sb08 Call-ID: ebeedfce-4be8-1232-63b0-00123f20329d Contact: <sip:66.33.147.149:5060> CSeq: 59143169 INVITE Server: sbc_3 Content-Type: application/sdp Content-Length: 152 v=0 o=SBCSIPUAS 1696166055 1 IN IP4 10.50.2.40 s=SBCSIPUAS SIP STACK v1.0 c=IN IP4 10.50.2.40 t=0 0 m=audio 23194 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ---------------------------------------------------------- aggressive-nat-detection [true] all-reg-options-ping [true] apply-inbound-acl [default] auth-calls [true] context [Inbound_Route] debug [info] dialplan [XML] disable-transcoding [true] disable-transfer [true] dtmf-duration [2000] enable-100rel [false] enable-timer [false] force-subscription-expires [60] ignore-183nosdp [no] inbound-bypass-media [true] inbound-codec-negotiation [generous] inbound-codec-prefs [G722,PCMU,PCMA,GSM] inbound-proxy-media [false] inbound-reg-force-matching-username [true] liberal-dtmf [true] manual-redirect [false] minimum-session-expires [60] outbound-codec-prefs [G722,PCMU,PCMA,GSM] registration-thread-frequency [30] rtp-end-port [42000] rtp-ip [10.178.239.32] rtp-start-port [ 16000] rtp-timeout-sec [60] session-timeout [60] sip-ip [10.178.239.32] sip-port [5060] sip-trace [false] tls [false] unregister-on-options-fail [true]

3 6

ldns error on install
by Joli Martinez 05 May '14

05 May '14

Hello I am trying to install freeswitch on CentOS and when I do "make install" i keep getting and error "*** You must install libldns-dev to build mod_enum. Stop." Any suggestions

9 14

Presence on park BLF keys
by Francis 05 May '14

05 May '14

Hi folks, I've got a couple of questions: 1. Do the default call park examples do BLF presence? I've tried adding @${domain} as per the wiki, but still no joy. 2. Is call_intercept capable of grabbing a call if it's on hold, as opposed to ringing? Cheers, Francis

2 3

FreeSWITCH -Install
by Andre 04 May '14

04 May '14

How do I make FreeSWITCH a service and add arguments? C:\ FreeSwitchConsole.exe -nosql -install FSService ? Thanks

2 1

Is FAXing reliable on FreeSwitch w/ T.38
by Sean Devoy 04 May '14

04 May '14

I have had some bad experiences trying to get reliable FAX support over VOIP (FS). However, I have not used T.38 capable ATAs. For the purposes of my questions please assume I am faxing FROM a fax machine over IP into the FS where I will "capture" it using <action application="rxfax" data="//my_directory//rxfax.tiff"/> Does a T.38 capable ATA improve the success rate of sending a fax from the ATA just into FreeSwitch? I have always thought T.38 was something you put on your Gateway Lines and I think I have forgotten my ATAs connections. Also, what success rate have you had with end-to-end delivery of FAXes using T.38? This accountant has heard FAXing is not reliable over VOIP. So far, I think I agree with him. Does T.38 really fix the problem? Thanks, Sean

5 10

possible to play streaming video
by Peter Villeneuve 03 May '14

03 May '14

Hi, I've been using mod_shout to stream audio to extensions and it works rather well. Is there a video version that would allow me to stream video to an extension? Thanks

1 0

Jira and the BugHunt!
by Ken Rice 02 May '14

02 May '14

Hey Guys, Jira is offline this morning for a bit so we can do a little maintenance on it. Also 2PM EST Join us on SIP:888@conference.freeswitch.org for the FridayBugHunt! Be there! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

1 1

Re: [Freeswitch-users] softswitch
by Dan Christian Bogos 02 May '14

02 May '14

Hi Hassan, Yes, I know the details of the configuration (done the benchmarks myself). Actually we are pretty serious with these benchmarks, by default CGRateS coming with it's own stress tool (cgr-tester binary) which you can use to benchmark your own rating profiles. The machine we have used is a virtual host with a one Intel 3.4 GHz CPU and 1GB of RAM. Hardisk does not matter since we do not store anything for calculations on HDD. All the data we use in rating calculations is cached using own cache mechanism. On this machine we got the following stats (querying cost for one emulated call repeatedly -100k times) : 1. No network involved in query, (querying internally within the same process): 163655 requests per second. Applicability: CGR-SM component directly attached to FreeSWITCH with internal connection to the rater. 2. Remote RPC using GOB as transport: 12501 requests per second. Applicability: CGR-SM running on the same host with FreeSWITCH, remote rater (eg: charging for a cluster of FreeSWITCHes). 3. Remote RPC using JSON-RPC as transport (worst case scenario): 6221 requests per second. Applicability: Third party written session manager using CGR-Rater component for it's rating and implementing it's own call handling logic. I should mention that the number of requests per second should not decrease with the number of destinations you search since we use individual hashes in our cache and tested so far with 6.5 Mil. destinations without impact on performance. Happy to go deeper in your questions, not sure if appropriate to abuse here the mailing list for them, otherwise you can always join our mailing list or irc channel for a more interactive conversation. DanB PS: For the sake of correctness the screenshoot of the performance benchmarks I am talking in this email is available to be directly sent to you if interested (not accepted by the mailing list because of size). On 02.05.2014 13:43, freeswitch-users-request(a)lists.freeswitch.org wrote: > CGRateS seems to be a very interesting project! > > The features page says "a machine of modest configuration" was used to > benchmark up to 13,000 req/sec. Would you happen to know details of this > "modest configuration"? > > Regards > HASSAN

2 1

Video-only call
by Richard Screene 02 May '14

02 May '14

Hello, Does FreeSWITCH support receiving calls that contain only a video component? (I am trying to use FreeSWITCH to distribute WebRTC screenshare streams to users. It works fine with audio and video) The INVITE looks like: INVITE sip:1273936153_290033102@a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356 max-forwards: 69 To: <sip:1273936153_290033102@a.b.c.d> From: "Rich" <sip:rscreene@gmail.com>;tag=mgu8fejfq8 call-id: llc4a6rbqb7886s43nhl CSeq: 7243 INVITE Contact: <sip:127.0.0.1:7071;transport=tcp> allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE supported: outbound user-agent: JsSIP 0.4.0-devel content-length: 1824 Content-Type: application/sdp v=0 o=- 4235676450597780712 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE video a=msid-semantic: WMS 1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd m=video 34116 RTP/SAVPF 100 116 117 c=IN IP4 e.f.g.h a=rtcp:34116 IN IP4 e.f.g.h a=candidate:2668419503 1 udp 2122260223 192.168.47.203 56323 typ host generation 0 a=candidate:2668419503 2 udp 2122260223 192.168.47.203 56323 typ host generation 0 a=candidate:3515819359 1 tcp 1518280447 192.168.47.203 0 typ host generation 0 a=candidate:3515819359 2 tcp 1518280447 192.168.47.203 0 typ host generation 0 a=candidate:1401022012 1 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0 a=candidate:1401022012 2 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0 a=candidate:3671437516 1 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0 a=candidate:3671437516 2 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0 a=ice-ufrag:vD/mqbgKKKvb+vwg a=ice-pwd:xbg5CyTfAIQMCD3JTmjU9IGw a=ice-options:google-ice a=fingerprint:sha-256 E8:6D:CF:75:43:C6:D9:C6:90:8E:FD:15:CD:FC:86:A2:90:E6:A3:BE:CF:A1:12:64:60:2F:26:89:96:C0:16:18 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=ssrc:2554093369 cname:L3M//yZhkStQCiOE a=ssrc:2554093369 msid:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd 908bd517-4202-4348-a36d-f474090f3cc2 a=ssrc:2554093369 mslabel:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd a=ssrc:2554093369 label:908bd517-4202-4348-a36d-f474090f3cc2 In the log I see the following message: 2014-05-02 13:35:43.267313 [DEBUG] switch_core_media.c:2223 No audio codec available And FreeSWITCH responds with: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport=7071;received=a.b.c.d Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356 Max-Forwards: 692014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene(a)gmail.com) State REPORTING From: "Rich" <sip:rscreene@gmail.com>;tag=mgu8fejfq8 2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:102 sofia/external/rscreene(a)gmail.com Standard REPORTING, cause: INCOMPATIBLE_DESTINATION To: <sip:1273936153_290033102@a.b.c.d>;tag=H3e25S58vjNZQ call-id: llc4a6rbqb7886s43nhl 2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene(a)gmail.com) State REPORTING going to sleep CSeq: 7243 INVITE User-Agent: Drum 1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "1273936153_290033102" <sip:1273936153_290033102@a.b.c.d>;party=calling;privacy=off;screen=no Do I have to configure something to allow video-only calls? Or, is it a limitation of FreeSWITCH? If it is a limitation of FreeSWITCH is it easy to remove? Many thanks, Richard Richard Screene Senior Developer Drum Web Meetings +44 1273 936125 www.thisisdrum.com Drum is the collaboration solution by NetDev Ltd. Registered in England and Wales Company Number 04741258

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