- FreeSWITCH-users - FreeSWITCH Mailing Lists

Incoming Fax problems with bad rows
by Raphael Lechner 04 Mar '15

04 Mar '15

Hi, We have a problem that some incomings fax have many bad rows, from at least 2 different customer and therefore not all lines are readable. We tried first with to remove the ATA Device and changed that, that we receive them by mod_spandsp but the problem still exists. Attached the log file: Sender 1:https://pastebin.freeswitch.org/23960 <https://pastebin.freeswitch.org/23960> Sender 2: https://pastebin.freeswitch.org/23961 <https://pastebin.freeswitch.org/23961> FreeSWITCH version: 1.4.15+git~20141229T185951Z~507a0f22c5~64bit (git 507a0f2 2014-12-29 18:59:51Z 64bit) We use a patton ISDN/VoIP Gateway. Any hint how we can resolve/debug that? Thank you, Raphael

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$${sounds_dir} variable
by Richard Mace 04 Mar '15

04 Mar '15

Hi all, Could someone please let me know which file contains the defined location for the variable $${sounds_dir} Thanks Richard

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Re: [Freeswitch-users] ICE SDP Params
by Rob Moore 04 Mar '15

04 Mar '15

Hi Brian, I thought as much, WebRTC isn’t something we are trying to use at the moment (although im sure we’ll find a use for it in the not too distant future.) Invites are created using the bridge application in XML dialplan. I have SIP pcaps but I don’t have and Freeswitch traces at the moment as the issue is only appearing once in say 500 calls on our production system so it can be a little awkward to pin down detailed tracing. I’m working on getting an example today and will post back as soon as possible. Is there any way to disable WebRTC entirely? That could be worth a try whilst I get a test setup for this scenario. From: Brian West <brian(a)freeswitch.org<mailto:brian@freeswitch.org>> Date: Mon, Mar 2, 2015 at 6:15 PM Subject: Re: [Freeswitch-users] ICE SDP Params To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> This is all related to WebRTC, how are you creating the invite? Logs would be helpful./b

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Load test new install
by Malay Thakershi 03 Mar '15

03 Mar '15

I just installed 64-bit FreeSwitch and want to move my 32-bit base to the new server. How can I do basic load test? How can I make dummy calls between two servers? Thanks.

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Getting FS to place Outbound calls
by T Fred Farmington 03 Mar '15

03 Mar '15

Your FreeSWITCH note came to me in 'digest' mode with other postings. So I think that this Reply will likely start a new thread when I hoped that it would not. So my apologies if this doesn't work as desired. Regardless, it seems like whenever I find a problem and report, I subsequently find a more basic problem which likely results in that which I reported. ================================================ Anyway, I still cannot get my Outbound calls to work. But at a more basic level, I find that when I do: sofia status I see that my gateway ( velocity-outbound ) is being reported as NOREG which I assume to be Not Registered and therefore Will Not Work. Name Type Data State ================================================================================ external-ipv6 profile sip:mod_sofia@[2002:6b01:26bd::6b01:26bd]:5080 RUNNING (0) external-ipv6::example.com gateway sip:joeuser@example.com NOREG external-ipv6::velocity-outbound gateway sip:FreeSWITCH@<SIP IP Here> NOREG <FS IP Here> alias internal ALIASED external profile sip:mod_sofia@<FS IP Here>:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::velocity-outbound gateway sip:FreeSWITCH@<SIP IP Here> NOREG internal-ipv6 profile sip:mod_sofia@[2002:6b01:26bd::6b01:26bd]:5060 RUNNING (0) internal profile sip:mod_sofia@<FS IP Here>:5060 RUNNING (0) ================================================================================ I contacted my SIP line vendor and ran a test. It seems that in spite of my using external-ipv6.xml they saw the INVITE come in from my FreeSWITCH and they responded with a 200 OK But they never saw anything else come back to them after that with which to complete the 'handshake' Interesting enough within external-ipv6.xml I modified it so that it would <gateways> would include nothing And within external.xml I added the <gateways> to include: <X-PRE-PROCESS cmd="include" data="external/outbound_via_velocity.xml"/> After which I did a full Restart on the FreeSWITCH service And in spite of that I still see in the freeswitch.log: Added gateway 'velocity-outbound' to profile 'external-ipv6' This is a total guess, but I assume that I need to see the velocity-outbound gateway that I am trying to use show up as a REG before I can go any further. If that is correct, how/where would I look to find out what is wrong? BTW: In the conf\dialplan\default directory I did set the other plan's extensions to .txt so that they would be ignored. Thanks TF

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FreeSWITCH 1.4.15 Released
by Ken Rice 03 Mar '15

03 Mar '15

New Post on freeswitch.org from krice387 check it out at http://freeswitch.org/freeswitch-1-4-15-released/ FreeSWITCH 1.4.15 Released FreeSWITCH 1.4.14 has been released! This is routine maintenance release. Source Tarball available at http://files.freeswitch.org/freeswitch-1.4.15.tar.bz2 Debian and Yum Repos have been updated as well. See the release notes below for a list of notable changes. Happy New Years From the FreeSWITCH Team! Release Notes: New features that were added: e55aee1 FS-7025 Add drop_dtmf_masking_tone channel_variable [Jira: https://jira.freeswitch.org/browse/FS-7025] a8c5a0c FS-7048 Add timezone support to mod_say_{de,es,ja,nl,th,zh} 17574a8 Add bert stats to mod_bert::lost_sync event a26e29c vs2010 support for recent unimrcp changes cee8b30 Set rtp_has_crypto for dtls calls 5fcff50 FS-7093 Create uuid_drop_dtmf [Jira: https://jira.freeswitch.org/browse/FS-7093] f024ea3 FS-7047 Arbitrary MRCP headers can now be sent to unimrcp input components in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7047] e783999 Some changes to webrtc to make it work with iDoubs in rtcweb profile mode d189e98 Allow 10ms jb 750b1dd FS-7114 Allow streaming binary data from mod_memcache Improvements in performance: 4bcf1d8 Use cached time to save cpu Improvements in cross platform build supports: 32c27b3 Added a Debian dependency to the CentOS6 makefile f4876d5 FS-7031 [unimrcp] update sofia-sip.m4 so that it can build when relative path is used in configure.gnu –with-sofia-sip [Jira: https://jira.freeswitch.org/browse/FS-7031] 061f3cb FS-7031 #resolve #comment [unimrcp] update library again to pull in upstream fix for –with-sofia-sip=../sofia-sip [Jira: https://jira.freeswitch.org/browse/FS-7031] 382e683 Use FTDM_UINT64_FMT macro to log uint64_t values, in order to not break x86 builds. dc9e904 FS-7025 fix compiler warning introduced from e55aee14 [Jira: https://jira.freeswitch.org/browse/FS-7025] b69c93e FS-7030 More work toward fixing FS build on Windows Visual Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030] db66cdb Fix mrcp libraries to build correctly c327455 FS-7030 More work toward getting FS to build on Windows Visual Studio 2012 [Jira: https://jira.freeswitch.org/browse/FS-7030] b341ff7 FS-7046: fix data types and casting on some vars to silence windows build warnings in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] 7ce5171 FS-7046 follow up on type change in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] 357ffad Fix windows build error 0b414a8 vs2010 unimrcp working build 0c1e698 Update build deps for debian list 0a0b926 Build fix for gcc 4.9 fixing a variable set but not used error in mod_commands af6b23a FS-7046 Fix some additional Windows build warnings for mod_verto [Jira: https://jira.freeswitch.org/browse/FS-7046] Additional documentation: f63f868 FS-7049 – Documentation for state optional paramenter in callcenter_config queue list and count [Jira: https://jira.freeswitch.org/browse/FS-7049] In terms of stability these were the use cases that were fixed: 392c687 FS-7055 Fix for a stability race condition in FS [Jira: https://jira.freeswitch.org/browse/FS-7055] d5119a7 FS-7091 Removed unnecessary mutex lock inside input component’s cleanup function since the input component won’t be cleaned up unless all references have been released, in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7091] These were the packaging improvements: 3c8dd3e Handle missing `lsb_release` 505cd29 Refactor distro detection and handling 430433a Improve error message d88bae1 Support optional debian parallel builds The following bugs were squashed: 5bbef7f FS-7015 Fix for inbound call on hold issue in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7015] 99a5b50 FS-7063 Fix for media delay issue [Jira: https://jira.freeswitch.org/browse/FS-7063] 21458f8 FS-7062 On redirect, when uri are passed in without <> with multiple uris, automatically add the q= header param in decending order in mod_sofia. [Jira: https://jira.freeswitch.org/browse/FS-7062] 5376e82 FS-6688 This will fix the normal case of record route from a proxy without breaking normal changing of a contact in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-6688] 06c241a FS-6891 FS-7002 FS-7059 FS-7072 FS-7073 FS-7076 #close #comment All of these bugs are invalidated due to a botched revert [Jira: https://jira.freeswitch.org/browse/FS-6891] 922fd81 FS-7015 The code was not properly catching the 0.0.0.0 after changing it to work with ICE SDPs because it was looking in the wrong place for the 0.0.0.0 [Jira: https://jira.freeswitch.org/browse/FS-7015] 3d515cf Re-mark cur_payload as negotiated when detected as such by parser or the rtp could stop working on session re-invite 19272dc FS-7078 Fix sip_header_as_string to properly null_terminate on larger header strings [Jira: https://jira.freeswitch.org/browse/FS-7078] e268a72 FS-6994 Fix for Codec OPUS decoder error in mod_opus [Jira: https://jira.freeswitch.org/browse/FS-6994] 6dbb416 FS-7086 FS-6798 Fix for invalid codec tearing down the call request [Jira: https://jira.freeswitch.org/browse/FS-7086] 46adbec FS-7030 #comment [unimrcp] restore visual studio 2010/2012 project files added by FS project [Jira: https://jira.freeswitch.org/browse/FS-7030] bad5dc3 FS-7037 Fix for T38 fax break started by commit 5bbef7f1e50 [Jira: https://jira.freeswitch.org/browse/FS-7037] 72c3df5 FS-6891 FS-6713 #comment revert 1b612fecb6e8db11da9b15c5522b87e7b642423d [Jira: https://jira.freeswitch.org/browse/FS-6891] 2a7b022 FS-6980 #resolve don’t crash when using native recording on recordstop the redo [Jira: https://jira.freeswitch.org/browse/FS-6980] 35ba6a3 FS-6766 Fix verto caller ringback missing on conference bridge in mod_verto [Jira: https://jira.freeswitch.org/browse/FS-6766] e8cf9c7 FS-7045 Guarantee that dialed call can be joined when answered event is sent in mod_rayo [Jira: https://jira.freeswitch.org/browse/FS-7045] 4be6290 FS-7052 Moving jb queue swap operation out of the debug block. [Jira: https://jira.freeswitch.org/browse/FS-7052] 843e495 FS-7051 Preserve the annexb=no/yes status in mod_sangoma_codec [Jira: https://jira.freeswitch.org/browse/FS-7051] 158c1f2 FS-7002 Fix for recorded audio being choppy when diferent ptimes present and record session starts on bleg [Jira: https://jira.freeswitch.org/browse/FS-7002] 4ce2ce3 FS-7092 Fixed bug with Comrex OPUS [Jira: https://jira.freeswitch.org/browse/FS-7092] d786490 Fix timestamps in mod_bert broken by the cpu improvements refactoring ba016c2 FS-7095 Fix for FS sending DTLS HELLO (and STUN binding request) to wrong port [Jira: https://jira.freeswitch.org/browse/FS-7095] e0dcd17 FS-7083 #comment patch to change mod_shout to use lame_encode_buffer_interleaved on stereo channels so we don’t have to mess with the input data [Jira: https://jira.freeswitch.org/browse/FS-7083] 326289c FS-7083 This patch adds a dedicated thread for writing to the file and the channel_variable RECORD_USE_THREAD=false will disable it and sync may still be maintained at the cost of dropping more data from the audio signal. [Jira: https://jira.freeswitch.org/browse/FS-7083] 9fabbab Disable hard-mute when a session has a media bug attached 0200bc1 FS-7083 Fix divide by zero [Jira: https://jira.freeswitch.org/browse/FS-7083] 067cb0f FS-7100 Make buffer for sub contact big enough in mod_sofia [Jira: https://jira.freeswitch.org/browse/FS-7100] 7798b2f FS-6984 Set default video rates [Jira: https://jira.freeswitch.org/browse/FS-6984] 763e6aa FS-7046 Fix warning introduced from b341ff7 [Jira: https://jira.freeswitch.org/browse/FS-7046] 65e678b FS-7070 Fix mod_expr `clamp` function typo 0a66db6 FS-7111 Fix for bridge_early_media crash [Jira: https://jira.freeswitch.org/browse/FS-7111] Miscellaneous commits: 0d636af FS-7031 [unimrcp] revert configure.gnu change- doesn’t work when using non-source build dir. [Jira: https://jira.freeswitch.org/browse/FS-7031]

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Fwd: How it works? Lua & MySQL throug ODBC
by Борисов, Дмитрий / Dmitriy Borisov 03 Mar '15

03 Mar '15

Hi, All! I have experienced periodicaly problems with FreeSWITCH running LUA scripts. This scripts are event hooks. In some unknown reasons some times FreeSWITCH crashes without any records in log. I`ve some qustions: 1. How can I enable more detailed debug? 2. How to store freeswitch.core in some explained previously place? 3. May be thread blocking while doing transaction to MySQL through ODBC the source of my problems? If yes, how to solve it? -- with best regards, Dmitriy Borisov

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Channels in the same session
by Juan Pablo L. 03 Mar '15

03 Mar '15

Hi, i m writing a module that needs to check for certain information in a database for the caller and the destination number, for this the module is subscribing to the CS_INIT channel events, so everytime a channel is created the module callback is called and it checks the numbers, the problem is that the callback gets called twice, for the creation of the a-leg of the call and the creation of the b-leg. Is there any way to accomplish what i m trying to do ? I have already try getting testing for the flags in the channel but it did not work, i might be doing it wrong maybe ? Thanks!

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inheritance parameters from sofia.conf.xml to sip_profiles
by Vladimir Getmanshchuk 03 Mar '15

03 Mar '15

Hi Everyone! Looks like inheritance of some parameters from sofia.conf.xml <global_settings> to sip_profiles does not work. I've faced to problem with next parameters which configured at <global_settings> in sofia.conf.xml: - rtp-autofix-timing - user-agent-string but has no effect. Please advice. -- Yours sincerely, Vladimir Getmanshchuk

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Determining if Freeswitch channel is using a video codec
by Dylan Mikus 03 Mar '15

03 Mar '15

I’m trying to determine if a given channel over Freeswitch is using a video codec. In my config/vars.xml file, I’ve set the codecs line to: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,H264"/> Logs My SDP negotiation appears to be correct. The INVITE: INVITE sip:queuecard@cyberdyne.onsip.com SIP/2.0 Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099 Max-Forwards: 70 To: <sip:queuecard@cyberdyne.onsip.com> From: "Bender Rodriguez" <sip:bender@cyberdyne.onsip.com>;tag=fneppn1lhh Call-ID: n3o4g4i724sq7qkekp07 CSeq: 8622 INVITE Proxy-Authorization: Digest algorithm=MD5, username="cyberdyne_bender", realm="jnctn.net", nonce="54f4ce2e000013e4888519dec3ca2ee1ef9023f82d4d8922", uri="sip:queuecard@cyberdyne.onsip.com", response="f842951ecc11c3510d1e1b7abcdeb51f", qop=auth, cnonce="d153n6udlh74", nc=00000001 Contact: <sip:1fjqm8ps@o8iatftbl1mn.invalid;transport=ws;ob> Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Content-Type: application/sdp Supported: 100rel,outbound User-Agent: SIP.js/0.6.3 InstaPhone Content-Length: 1649 v=0 o=Mozilla-SIPUA-35.0.1 10886 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 a=ice-ufrag:a256418b a=ice-pwd:62e2ae7154b57f00ed0b1a2003ccf7af a=fingerprint:sha-256 EA:C4:92:D4:94:62:18:41:39:2E:42:B4:4E:B7:32:9E:66:FE:7C:01:57:AC:2C:4C:E4:66:4F:3B:B6:91:FA:DC m=audio 9 UDP/TLS/RTP/SAVPF 109 9 0 8 101 c=IN IP4 0.0.0.0 a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2130379007 192.168.1.38 58531 typ host a=candidate:0 2 UDP 2130379006 192.168.1.38 64677 typ host a=candidate:1 1 UDP 1694236671 38.104.167.182 49209 typ srflx raddr 192.168.1.38 rport 58531 a=candidate:1 2 UDP 1694236670 38.104.167.182 51209 typ srflx raddr 192.168.1.38 rport 64677 m=video 9 UDP/TLS/RTP/SAVPF 120 126 97 c=IN IP4 0.0.0.0 a=rtpmap:120 VP8/90000 a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42e01f;packetization-mode=1 a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=42e01f a=sendrecv a=rtcp-fb:120 nack a=rtcp-fb:120 nack pli a=rtcp-fb:120 ccm fir a=rtcp-fb:126 nack a=rtcp-fb:126 nack pli a=rtcp-fb:126 ccm fir a=rtcp-fb:97 nack a=rtcp-fb:97 nack pli a=rtcp-fb:97 ccm fir a=setup:actpass a=rtcp-mux a=candidate:0 1 UDP 2130379007 192.168.1.38 59562 typ host a=candidate:0 2 UDP 2130379006 192.168.1.38 61464 typ host a=candidate:1 1 UDP 1694236671 38.104.167.182 59357 typ srflx raddr 192.168.1.38 rport 59562 a=candidate:1 2 UDP 1694236670 38.104.167.182 21168 typ srflx raddr 192.168.1.38 rport 61464 The 200 OK response: SIP/2.0 200 OK Via: SIP/2.0/WS o8iatftbl1mn.invalid;branch=z9hG4bK8962099 Record-Route: <sip:38.109.82.167;lr;ftag=fneppn1lhh;did=24.0b0068f6> Record-Route: <sip:38.109.82.192;lr;ftag=fneppn1lhh;did=24.4c4e8343> Record-Route: <sip:38.109.82.193:5060;transport=udp;lr;ovid=cac1e79d> Record-Route: <sip:2b14c410c1@38.109.82.193:80;transport=ws;lr;ovid=cac1e79d> From: "Bender Rodriguez" <sip:bender@cyberdyne.onsip.com>;tag=fneppn1lhh To: <sip:queuecard@cyberdyne.onsip.com>;tag=30yQvF62DQyyg Call-ID: n3o4g4i724sq7qkekp07 CSeq: 8622 INVITE Contact: <sip:dtlsgw0@app2-1.55-broad-1.app.jnctn.net> Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, PRACK, NOTIFY Supported: precondition, 100rel, timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 1738 v=0 o=FreeSWITCH 1425307798 1425307799 IN IP4 38.109.82.228 s=FreeSWITCH c=IN IP4 38.109.82.228 t=0 0 a=msid-semantic: WMS 61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw m=audio 21882 UDP/TLS/RTP/SAVPF 109 101 a=rtpmap:109 opus/48000/2 a=fmtp:109 useinbandfec=1;usedtx=1;maxaveragebitrate=30000 a=rtpmap:101 telephone-event/8000 a=recvonly a=silenceSupp:off - - - - a=ptime:20 a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2 a=rtcp:21883 IN IP4 38.109.82.228 a=ssrc:2365215248 cname:CkzZ9cdxFymMTiha a=ssrc:2365215248 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a0 a=ssrc:2365215248 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a=ssrc:2365215248 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwa0 a=ice-ufrag:zqC4ZTWsD5d4Hyqa a=ice-pwd:IQ0IT35osh0bq7bPoDKenwwR a=candidate:9358589392 1 udp 659136 38.109.82.228 21882 typ host generation 0 a=candidate:9358589392 2 udp 659134 38.109.82.228 21883 typ host generation 0 m=video 23680 UDP/TLS/RTP/SAVPF 126 b=AS:256 a=rtpmap:126 H264/90000 a=fmtp:126 profile-level-id=42e01f;packetization-mode=1 a=recvonly a=fingerprint:sha-256 D6:87:51:92:F6:80:BE:0D:5B:9A:97:C3:53:A7:40:C5:A2:19:60:CA:48:2D:18:A2:53:AF:B6:E1:4E:02:39:D2 a=rtcp:23681 IN IP4 38.109.82.228 a=rtcp-fb:* fir pli a=ssrc:1652571152 cname:CkzZ9cdxFymMTiha a=ssrc:1652571152 msid:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw v0 a=ssrc:1652571152 mslabel:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvw a=ssrc:1652571152 label:61ZdxYMnyCuYhu6FrqjCFIr1DUjUALvwv0 a=ice-ufrag:kirEYQCPyInSpi7Y a=ice-pwd:fbIhKWJB3fFuGVyQ4QlSwNxU a=candidate:9055446981 1 udp 659136 38.109.82.228 23680 typ host generation 0 a=candidate:9055446981 2 udp 659134 38.109.82.228 23681 typ host generation 0 We offer: a=rtpmap:120 VP8/90000 a=rtpmap:126 H264/90000 and we accept: a=rtpmap:126 H264/90000 Note that this is on Firefox 35.0.1. Response While this call is up, I run show channels in fs_cli and get the following: uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num b41772a4-c11b-11e4-a78f-7585ed98a76c,inbound,2015-03-02 20:35:45,1425328545,sofia/sip0/bender@cyberdyne.onsip.com,CS_SOFT_EXECUTE,Bender Rodriguez,bender,38.109.82.167,queuecard,uuid_bridge,bc7c6d64-c11b-11e4-a798-7585ed98a76c,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Outbound Call,terabithia bc7c6d64-c11b-11e4-a798-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/app/c3po@cyberdyne.onsip.com,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,1000,uuid_bridge,b41772a4-c11b-11e4-a78f-7585ed98a76c,XML,generic-app,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,terabithia,SEND,b41772a4-c11b-11e4-a78f-7585ed98a76c,Bender Rodriguez,bender bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,inbound,2015-03-02 20:35:59,1425328559,sofia/sip0/bender@cyberdyne.onsip.com,CS_EXECUTE,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,bridge,{force_transfer_context=refer}sofia/sip0/sip:gl8k15o7@e6kin9qicasi.invalid;transport=ws;aor=c3po%40cyberdyne.onsip.com,XML,default,opus,48000,0,opus,48000,0,,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Outbound Call,gl8k15o7 bc82805a-c11b-11e4-a7ae-7585ed98a76c,outbound,2015-03-02 20:35:59,1425328559,sofia/sip0/sip:gl8k15o7@e6kin9qicasi.invalid,CS_EXCHANGE_MEDIA,Bender Rodriguez,bender,38.109.82.167,gl8k15o7,,,XML,default,opus,48000,0,opus,48000,0,srtp:dtls:AES_CM_128_HMAC_SHA1_80,app-server2-1.55-broad-1.jnctn.net,,,ACTIVE,Outbound Call,gl8k15o7,SEND,bc80f1f4-c11b-11e4-a7a0-7585ed98a76c,Bender Rodriguez,bender The read codecs and write codecs are OPUS, except for a websocket transport that lists an XML codec, I think. Is something up with my setup, or do we only show the audio codec being used when we run the show channels command? Any other idea for how to determine whether a Freeswitch channel is using video? I’m trying to stay away from sending custom headers and I want to be able to figure this out within Freeswitch. In other words, I don’t want a receiving application to try to figure out whether it is in video or not. I just want to query my Freeswitch service to find out. Thanks, guys! I appreciate any help. -- Dylan Mikus Software Engineer OnSIP www.onsip.com p. 212.933.9190 x7060 SIP/Email: dylan(a)onsip.com

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