May 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

voicemail_ivr examples when checking voicemail
by Jonathan Hunter 10 May '17

10 May '17

Hey guys, Sorry for the noise! I am looking to use voicemail_ivr so I can ensure my own phrases are played and I can avoid giving the user options to record just one voicemail message (as they have the option to upload a file from our portal too). Can this be achieved using voicemail_ivr and if so are there any examples as confluence is good but doesnt have many configuration examples if you want to miss out phrases etc. Cheers Jon

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Re: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference
by Zach Rothy 10 May '17

10 May '17

So I tested with Freeswitch 1.6.17 as Brian suggested earlier, and it seems that the errors and warnings I saw still occur. To summarize the original post I made, I have my Freeswitch server which supplies the server for video conferences using Verto. I have it bridging to an Asterisk server (testing with Asterisk 13) to allow PSTN calls to be bridged to Freeswitch via an Asterisk MeetMe conference room. Freeswitch gets its dialplan for Verto via mod_xml_curl and sends the call to Asterisk via the conference auto outcall application. When I set the flags for this bridged call to ghost it seems Freeswitch is receiving the RTP, as I see verto say "Talking" properly, but I never hear anything from my verto client connected to the video conference room. This does not happen when the ghost flag is removed, as I can speak from either end and hear it on the other one. Additionally when the bridged call initially joins the conference, it seems to be being placed as the presenter even though I have the flags so that without video it can't be on the canvas. Such I get the following warning flooding the console: 2017-05-10 12:51:57.505225 [WARNING] switch_core_media.c:11774 sofia/internal/352650@asterskdev has no video codec This warning stops both when the floor is not trying to be set to the audio bridge, or when the floor is locked to another presenter. It recognizes properly you can't set the vid floor to that bridged call in fs_cli as it says in console: 2017-05-10 13:33:18.125230 [DEBUG] conference_event.c:110 conf 3525236 CMD mod_verto [vid-floor] 0002 2017-05-10 13:33:18.125230 [ERR] conference_api.c:1770 Channel sofia/internal/352650@hasterisk-dev does not have video capability! 2017-05-10 13:33:18.125230 [ALERT] conference_event.c:316 RES [3525236 vid-floor 0002 force][] I've included both the profile and dialplan returned by my web application for mod_xml_curl. Dialplan: <document type="freeswitch/xml"> <section name="configuration"> <configuration name="conference.conf" description="Conf Profiles"> <profiles> <profile name="3525236"> <param name="domain" value="$${domain}"/> <param name="rate" value="48000"/> <param name="channels" value="2"/> <param name="interval" value="20"/> <param name="energy-level" value="200"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="hold-music" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|livearray-json-status|video-required-for-canvas|video-mute-exit-canvas"/> <param name="video-mode" value="passthrough"/> <param name="video-layout-name" value="1x1"/> <param name="video-layout-name" value="group:grid"/> <param name="video-canvas-size" value="1920x1080"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-fps" value="15"/> <param name="max-members-sound" value="conference/conf-conference_is_full.wav"/> <param name="endconf-grace-time" value="30"/> <param name="max-members" value="30"/> </profile> </profiles> </configuration> </section> </document> Dialplan(Returned from mod_xml_curl): <document type="freeswitch/xml"> <section name="dialplan" description="Dialplan routing"> <context name="default"> <extension name="3525236"> <condition field="destination_number" expression="^3525236-moderator$"> <action application="set" data="api_hangup_hook=system ROOM=3525236 VCONFID=239 /usr/local/bin/flip_conference_flag.sh"/> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_utils_auto_outcall_flags=ghost"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=3525236"/> <action application="conference_set_auto_outcall" data="sofia/internal/352650@asterisk_server_hostname"/> <action application="sleep" data="2000"/> <action application="conference" data="3525236@3525236+flags{moderator|endconf}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document>

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OPUS Codec
by Colton Conor 10 May '17

10 May '17

Does freeswitch support transcoding from OPUS to G711? How well does this work in theory? Having full HD come in via OPUS from the phone, and then transcoding to crappy G711. Does it sound funny? We are thinking that OPUS has superior audio quality, and can withstand up to 30 percent packet loss without any audio distorition. However, I would say the majority of our users calls as still to the PSTN which only speaks G711. How much processing power would it take to transcode from OPUS to G711? How well is the OPUS codec implemented on newer IP phones? I hear the Yealink supports OPUS with their new S line of phones, but I also heard they pulled support for it? Polycom seems to only support it on the VVX 500 and VVX 600, but you have to disable video according to the 5.4 release notes. OPUS must be processor intensive I assume if it can't run OPUS and video at the same time. Have any idea if the newer VVX, like the 411, 501, and 601 have this same limitation? I know they have faster processors and ram.

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Re: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference
by Zach Rothy 10 May '17

10 May '17

> You should probably give it a shot on newest release which is 1.6.17. I'll check out the newer version of freeswitch to see if it solves the issue. > Why would you want to bridge another conference in, instead of just > running the single conference? Doing so will allow you full access to > all the controls for individual members. You could still do this back > to back with freeswitch and asterisk by changing asterisk dial plan to > bridge to the conference room on freeswitch for each member. I have it set up so Freeswitch sends it to Asterisk instead of the other way around, as the Asterisk audio bridges are a completely separate feature. Bridging Freeswitch to Asterisk gives them the option of dialing into a video conference for the audio portion without having to change infrastructure a decent amount so that selectively some numbers go to Freeswitch and others Asterisk. I don't think it would be approved anyway to do that, partially due to infrastructure /routing changes for multiple servers, as well as the way video conference is billed versus the Asterisk conference rooms.

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[NOTICE] sofia.c: [INCOMPATIBLE DESTINATION] (setting up NAPTR records?)
by Ravi sanyal 10 May '17

10 May '17

This is a follow up to the previous problem. I was able to fix the certificates and stop the codec error, however it's still getting INCOMPATIBLE DESTINATION and hanging up as a result. the debug log looks like this: 2017-05-10 14:40:55.164862 [DEBUG] sofia.c:7257 Remote SDP: v=0 o=- 7420685792921982190 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE data a=msid-semantic: WMS m=application 55446 DTLS/SCTP 5000 c=IN IP4 111.69.178.179 a=candidate:1520298561 1 udp 2113937151 192.168.178.32 55446 typ host generation 0 network-cost 50 a=candidate:2664932878 1 udp 2113939711 2406:e007:2f57:1:1d0a:7773:e5e:f3d4 55447 typ host generation 0 network-cost 50 a=candidate:842163049 1 udp 1677729535 111.69.178.179 55446 typ srflx raddr 192.168.178.32 rport 55446 generation 0 network-cost 50 a=ice-ufrag:cCqv a=ice-pwd:m1PW5eEuaLQiFndNoj8Fm9q5 a=fingerprint:sha-256 AB:05:44:B9:4E:AE:4E:FA:E1:74:DE:0B:8C:41:B4:68:42:96:48:B4:C8:46:5A:20:F9:3B:7A:A8:AC:84:64:8E a=setup:actpass a=mid:data a=sctpmap:5000 webrtc-datachannel 1024 2017-05-10 14:40:55.164862 [NOTICE] sofia.c:7728 Hangup sofia/internal/ 1001(a)chat.fleetchat.co.nz [CS_NEW] [INCOMPATIBLE_DESTINATION] The sip trace looks like this: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/WSS 21d9a8odj6h4.invalid;branch=z9hG4bK3506982;received=111.69.178.179;rport=57055 Max-Forwards: 70 From: <sip:1001@chat.fleetchat.co.nz>;tag=400sjnc8e6 To: <sip:1002@chat.fleetchat.co.nz>;tag=Bta6SDNmvt28j Call-ID: 6c5ktogfdljrhk3odh8c CSeq: 7347 INVITE User-Agent: FreeSWITCH-mod_sofia/1.9.0-380-c66a012~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "1002" <sip:1002@chat.fleetchat.co.nz >;party=calling;privacy=off;screen=no paste bin of extended sip trace: https://pastebin.freeswitch.org/view/304d4f82 I think it's something to do with SCTP, but knowing nothing of this standard, I am not sure if this is the problem i'm facing. I'm guessing it may be not getting the required information between, since if it is trying to send through sctp ports, my server is not currently set up for that. So how do you set up sctp on freeswitch? i have an article im looking at: https://wiki.freeswitch.org/wiki/SIP_TLS which mentions NAPTR and SRV records. Do you need to set up a DNS server to make NAPTR records? Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

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Re: [Freeswitch-users] Audio Bridge to Asterisk from Freeswitch video conference
by Zach Rothy 09 May '17

09 May '17

Good Afternoon, I forgot to mention I am currently running Freeswitch 1.6.12 64bit on Centos 7 from the Freeswitch RPM repo. Thanks in advance, Zach

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Audio Bridge to Asterisk from Freeswitch video conference
by Zach Rothy 09 May '17

09 May '17

Good Afternoon, I currently have a video conference which users connect to via the Verto library. I was looking to enhance this conference by also allowing users to join audio-wise to the video conference by making a bridge/call to my asterisk server which hosts audio conference rooms. Currently I have the conference_set_auto_outcall making the call via sip to the asterisk server, but ran into a couple of issues after asterisk and the freeswitch servers establish the call. The first issue, though it may be on purpose, is that when I use the ghost flag for the member it seems RTP is never delivered to my verto endpoint from the sip call to the asterisk as well as the other way around. Though I clearly see freeswitch is recognizing energy detection from my asterisk conference, as I see the talking flag appear but no audio is heard from it on the verto client. It is the same the other way around, where I will talk via a mic on my verto client, but never hear it on the asterisk side of things. Though for asterisk it does not even seem to be sending audio to the asterisk server. I'd prefer to use the ghost flag as I do with screen sharing so that the audio bridge does not count as a user for the video conference, of course I can always get around this by just bumping up the max users by one too if the user decides to use the bridge. The second issue might be more of a misconfiguration issue, but I am a bit stuck on where to check. For other times when I don't have the ghost or any flag set for the outbound bridge call, things work as expected. I can transmit and hear audio from both ends to the other one without any issue. Though even though Freeswitch knows the bridged call has no video since has_video is set to false in conference <Room> xml_list, it seems to still try to use that as the presenter / floor occasionally. During these times I get the following in fs_cli constantly 2017-05-09 17:10:05.742387 [WARNING] switch_core_media.c:11459 sofia/internal/352650@asterisk-server has no video codec wher asterisk server is actually the IP/dns host name of the asterisk server I develop with. Ideally if I were not to use the ghost flag I'd want it to treat this bridged call as audio only and not constantly spew that warning message. Some ideas I haven't tested yet as I am not too familiar with freeswitch besides verto/mod_conference a bit, was to see if I could have it join the conference with an audio only profile maybe, or somehow set the SIP not to try offering video maybe. Any ideas? Here is the profile and dialplan I was using to test Conference profile (returned from mod_xml_curl): <document type="freeswitch/xml"> <section name="configuration"> <configuration name="conference.conf" description="Conf Profiles"> <profiles> <profile name="3525236"> <param name="domain" value="$${domain}"/> <param name="rate" value="48000"/> <param name="channels" value="2"/> <param name="interval" value="20"/> <param name="energy-level" value="200"/> <param name="muted-sound" value="conference/conf-muted.wav"/> <param name="unmuted-sound" value="conference/conf-unmuted.wav"/> <param name="alone-sound" value="conference/conf-alone.wav"/> <param name="hold-music" value="$${hold_music}"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="kicked-sound" value="conference/conf-kicked.wav"/> <param name="locked-sound" value="conference/conf-locked.wav"/> <param name="is-locked-sound" value="conference/conf-is-locked.wav"/> <param name="is-unlocked-sound" value="conference/conf-is-unlocked.wav"/> <param name="pin-sound" value="conference/conf-pin.wav"/> <param name="bad-pin-sound" value="conference/conf-bad-pin.wav"/> <param name="caller-id-name" value="$${outbound_caller_name}"/> <param name="caller-id-number" value="$${outbound_caller_id}"/> <param name="comfort-noise" value="false"/> <param name="conference-flags" value="video-floor-only|rfc-4579|livearray-sync|minimize-video-encoding|livearray-json-status|video-required-for-canvas|video-mute-exit-canvas"/> <param name="video-mode" value="passthrough"/> <param name="video-layout-name" value="1x1"/> <param name="video-layout-name" value="group:grid"/> <param name="video-canvas-size" value="1920x1080"/> <param name="video-canvas-bgcolor" value="#333333"/> <param name="video-layout-bgcolor" value="#000000"/> <param name="video-codec-bandwidth" value="1mb"/> <param name="video-fps" value="15"/> <param name="max-members-sound" value="conference/conf-conference_is_full.wav"/> <param name="endconf-grace-time" value="30"/> <param name="max-members" value="30"/> </profile> </profiles> </configuration> </section> </document> Dialplan(Returned from mod_xml_curl): <document type="freeswitch/xml"> <section name="dialplan" description="Dialplan routing"> <context name="default"> <extension name="3525236"> <condition field="destination_number" expression="^3525236-moderator$"> <action application="set" data="api_hangup_hook=system ROOM=3525236 VCONFID=239 /usr/local/bin/flip_conference_flag.sh"/> <action application="answer"/> <action application="set" data="conference_auto_outcall_timeout=5"/> <action application="set" data="conference_utils_auto_outcall_flags=ghost"/> <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> <action application="set" data="conference_auto_outcall_profile=3525236"/> <action application="conference_set_auto_outcall" data="sofia/internal/352650@asterisk_server_hostname"/> <action application="sleep" data="2000"/> <action application="conference" data="3525236@3525236+flags{moderator|endconf}"/> <action application="hangup"/> </condition> </extension> </context> </section> </document> Thanks in advance, Zach

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No events after a hangup
by Aníbal César Aguiar de Carvalho 09 May '17

09 May '17

Hi everyone. I have a scenario with the following flow: - Freeswitch is configured to forward connections to a determined number to a localhost port and my software is connected at this socket. - A call is made by a first user to this number. - This call is connected and answered by our software. - A second channel is originated to a second user, with the origination_caller_id_number of the first user. At this point if the second user answers or rejects the call (manually or due to a connection timeout) then the originate event, answer or hangup and others events are sent by Freeswitch, If the first user hangs up, no event is sent whatsoever. Is this how Freeswitch is supposed to work, or is it a bug? 'event plain ALL' is used at the beginning, so it is not a filter issue from my side. Any help is appreciated. Thanks, -- Aníbal Carvalho. anibal.carvalho(a)gmail.com anibal.carvalho(a)vulcanet.com.br

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Noisy logs (switch_rtp.c: 1086 missed 1)
by Allan Kristensen 09 May '17

09 May '17

Hello, I'm getting thousand of these every hour on my Freeswitch log: May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 08:09:24.807956 [WARNING] switch_rtp.c:1086 missed 1 May 8 08:09:32 sip1b media-switch-container[18466]: 2017-05-08 08:09:25.787957 [WARNING] switch_rtp.c:1086 missed 1 : I can see it's related to calls coming from an Asterisk server but the INVITE's look fine to me and calls are active. The customer says there are no problems, but it's making my log monitor unhappy (they are warnings after all). I've looked into the code and I can see it's something to do with handling of ICE, but why it appears is a little mysterious to me. (I using "apply-candidate-acl = wan.auto", so nothing fancy there). Any clues? /Allan

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[ERR] mod_sofia.c:2491 CODEC NEGOTIATION ERROR.
by Ravi sanyal 09 May '17

09 May '17

I'm sorry for pasting the whole code in the email. Here's a much more succinct version. pastebin of code is included below. I'm new to using freeswitch and need to set up so that i can use a webrtc data channel to connect two clients together. I'm using SIPJS. I followed the guide to setting up webrtc on freeswitch end, however to begin with it gave a 488 error and INCOMPATIBLE_DESTINATION. Finding out about this led me to codec negotation (the logs showed that when sending the invite from client A produced a CODEC NEGOTATION ERROR) Then I followed a codec negotation guide on the confluence page, however I'm new to freeswitch. The guide talked about outbound and inbound, however it then talked about a profile for outbound. I have an internal and external profile, and was setting the variables there. Having set inbound-late-negotation and inbound-zrtp-passthru and not finding the others in the profile, i restarted freeswitch to get this error: (on both internal and external profile) i'll give you the whole sip trace as it seems important. It would also be helpful to know what the internal profile is doing in terms of webrtc. the guide says that internal is only used internally. As in the local network. Im not connected to freeswitch locally. what is the purpose of the internal profile then? It was at points getting 503 service unavailable but it's gone back to 488. I'm not sure what the problem is. https://pastebin.freeswitch.org/view/ca875188 Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

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