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Verto call error
by Agustí Ubalde 26 Apr '17

26 Apr '17

Hi all, I'm testing a simple web page with Verto Communicator according with the guide http://evoluxbr.github.io/verto-docs/tut/making-a-call.html but it's not working. After making the call the following errors showed in the browser console: *Uncaught TypeError: Cannot read property 'srcObject' of undefined* * at onRemoteStream (verto-min.js:28)* * at Object.self.peer.RTCPeerConnection.onRemoteStream (verto-min.js:58)* * at RTCPeerConnection.peer.onaddstream (verto-min.js:70)* *Uncaught TypeError: Cannot read property 'sinkId' of undefined* * at $.verto.dialog.setAudioPlaybackDevice (verto-min.js:263)* * at verto-min.js:271* Any idea? Thanks, PRESENCE TECHNOLOGY An ENGHOUSE INTERACTIVE Company Agustí Ubalde Bellot Chief Developer C/ Comte Urgell 240 3º-A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 322 Enghouse is listed on the Toronto Stock Exchange, stock symbol: ENGH -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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Downgrade with Deb packages
by Emrah 26 Apr '17

26 Apr '17

Hello all, Can a good soul point me to the procedure to downgrade from the master branch to the stable branch using deb packages? Replacing unstable with stable and pointing to version 1.6 in my sources.list, followed by the update upgrade or dist-upgrade doesn't do the trick. Thanks!

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fs_encode opus to wav is it possible
by Antonio Silva 26 Apr '17

26 Apr '17

Hi, i'm trying to get rtp opus to a wav file using fs_encode but is not working... anyone have tried to decode rtp with opus? can it be done with fs_encode? So for what i did was: - wireshark -> rtp streams -> save as raw format - fs_encode -v -l mod_opus 0x60f26a47.opus audio.wav fs_encode creates the audio.wav but is empty.. i've done this with g729 files but with opus is not working... You can check the output of the command in https://pastebin.freeswitch.org/view/9cedcf56 Thanks for the help. -- Saludos / Regards / Cumprimentos, António silva

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FSClient and sip_auto_answer
by Paul Mateer 26 Apr '17

26 Apr '17

Hi folks. I have a FreeSWITCH server with a couple of FSClient softphones attached. I've been experimenting with conference calls using the following extension configuration (based upon the Mad Boss example): <extension name="Broadcast Announcement"> <condition field="destination_number" expression="^0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Broadcast"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_utils_auto_outcall_flags=mute"/> <action application="set" data="end-conf-grace-time=0"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="user/5006@$${domain}"/> <action application="conference_set_auto_outcall" data="user/5007@$${domain}"/> <action application="conference" data="SystemBroadcast@default++flags{deaf|moderator|endconf}"/> </condition> </extension> I've noticed a couple of issues which seem to be tied to the sip_auto_answer setting. 1. The phone making the call is providing a live radio feed, so the callees should receive audio, but they do not 2. When the caller disconnects from the conference, the callees remain connected despite the fact that the moderator and endconf flags are set. If I change the sip_auto_answer setting from true to false and then place the call, each phone rings until answered but 1. Once a phone is answered you hear the radio feed. 2. When the caller disconnects, so do all the callees. Can anyone explain why I am seeing the difference in behaviour between the true and false settings for sip_auto_answer? I was expecting the same behaviour in both cases and that setting the sip_auto_answer flag would simply mean that I didn't have to explicitly answer each phone added to the conference. Paul

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Choppy audio when conferencing 4+ participants
by Bilal Dar 26 Apr '17

26 Apr '17

I have been struggling with an issue for almost 2 weeks. Our regular calls have no quality issue and looking RTCP statistics network conditions are perfect. We have normally on peak hr 60 calls and around 10 conferences. We have noticed that when we have 2 conferences of 4 or 5 participants, audio starts breaking for the users who are on conference. Regular calls do not experience any quality degradation. I upgraded the server to specs of 30Gig memory and 8 vCPU but still the issue exists. Common thing I have noticed even during off-peak hrs is that two 4+ participant call can cause the issue. I have ruled out network & hardware. Last change I made was moved all users to G.711 from G.722. Now I am not sure what other steps I can take. Appreciate any suggestions. Thanks

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Voicemail callback option 5 disable
by manish thakor 25 Apr '17

25 Apr '17

Hello, Can we disable callback option 5 in voicemail?

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mod_voicemail - Voicemail check - choose greeting between 1 and 9-Disable
by Jonathan Hunter 25 Apr '17

25 Apr '17

Hi All, I was wondering if someone could advise if this is possible? When checking voicemail, and selecting record a greeting in advanced options using the default voicemail profile, it then prompts to choose a greeting between 1 and 9 and then a related .wav file is created, ie greeting_1.wav greeting_9.wav etc. With the default settings is it possible to* disable* that prompt with configuration, and also specify the name of the file I want creating, ie call it voicemail.wav instead of greeting_X.wav? Or am I better using a custom lua script or something else? Many thanks Jon

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Strange problem when using group_call
by Paul Mateer 25 Apr '17

25 Apr '17

I've been doing a little playing around with conference calls and I've encountered some odd behaviour that perhaps someone can account for. I have a system with FreeSWITCH version 1.7 along with the FSClient (Client Id 5006). I have another system running FSClient (ClientId 5007). I have the following extension added in my public dialplan: <extension name="group_dial_surveillance"> <condition field="destination_number" expression="^1000$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="originate_continue_on_timeout=true"/> <action application="set" data="call_timeout=0"/> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data="${group_call(surveillance@${domain_name}+A)}"/> <action application="transfer" data="1000 XML default"/> <action application="hangup"/> </condition> </extension> If I dial the number 1000 from a third softphone, then both FSClient phones ring and when one answers the call the other stops ringing. No problem there - that's exactly what I would expect as the bridge application is being passed the output for a group_call on the surveillance group (consisting of users 5006 and 5007). I also have the following Mad Boss extension in my dialplan: <extension name="Mad Boss"> <condition field="destination_number" expression="^0911$"> <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/> <action application="set" data="conference_auto_outcall_caller_id_number=0911"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="end-conf-grace-time=1"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(surveillance@${domain_name}+A)}"/>  <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/> </condition> </extension> The problem is that when the third softphone is used to dial 0911, only one of the two phones answers. Looking through the log files I've found that when 1000 is dialled and the group_dial_surveillance extension is processed the following is logged: EXECUTE sofia/internal/4008(a)10.51.54.185 bridge([^^:sip_invite_domain=10.51.54.76:presence_id=5006@10.51.54.76]sofia/internal/sip:gw+1@10.51.54.76:12346;transport=udp;gw=1,[^^:sip_invite_domain=10.51.54.76:presence_id=5007@10.51.54.76]sofia/internal/sip:gw+3@10.51.54.185:12346;transport=udp;gw=3) but when 0911 is dialled and the Mad Boss extension is executed the following is logged: EXECUTE sofia/internal/4008(a)10.51.54.185 conference_set_auto_outcall([^^:sip_invite_domain=10.51.54.76:presence_id=5006@10.51.54.76]sofia/internal/sip:gw+1@10.51.54.76:12346;transport=udp;gw=1,[^^:sip_invite_domain=10.51.54.76:presence_id=5007@10.51.54.76]error/user_not_registered) As far as I can see the data passed to the bridge and conference_set_auto_outcall should be the same - the data entries for both are identical, but in the case of the conference call for some reason the group_call is not locating the second user. If I switch to explicitly listing the users instead of using the group_call both phones will ring, if I switch back to using group_call then the second phone doesn't ring. and the same user_not_registered message is logged. It's completely reproducible as every call to 1000 works every time and every call to 0911 fails every time. As mentioned, I can work around the issue, but I'm kind of curios as to what would be causing it. Paul PS. On a slightly different note, is there a way to terminate the conference when the person who initiates it (the moderator?) hangs up?

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Sofia Profile to use for dialing gateway
by Lợi Đặng 25 Apr '17

25 Apr '17

Hi guys, I'm going to use FreeSWITH with 2 SIP Profiles, called internal and external. When calls come to external profile (port 5080), I'm dialing out a pre-defined gateway with bridge *<action application="bridge" data="sofia/gateway/${gateway}/${destination_number}"/>* Call is normal, except FreeSWITCH used internal profile (port 5060) contacting my gateway. I saw that internal profile is listed before external profile in *profiles *configuration, so I guess internal is considered higher priority or default. Calls come to internal profile have normal behavior, because FS use internal for dialing out. For some reasons, I'm not using user directory or REGISTER at all. So, can I choose a specific SIP Profile to use for dialing out via a gateway? just like dialing to *sofia/profile_name/destination *? Sorry if this question is duplicated elsewhere, any help would be appreciated. rgds, Loi Dang

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Park & Retrive call with different codec in call legs
by Donny Hardyanto 25 Apr '17

25 Apr '17

Hi, I tried to do valet parking, with first call (from gsm handphone using g711) parked, and retrieved by second call (from webrtc/webpage using opus). The park and retrieve is success, but the sound only one way. the gsm handphone can hear the user from webpage, but the user in webpage cannot hear the gsm handphone user. Is it a problem when using park/retrieve when using difference codec? Donny

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