July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

group_call_function and record_session - Recording calls for a specific Extension within a call group
by Jonathan Hunter 20 Jul '17

20 Jul '17

Hi All, Hope everyone is well? I have a quick question around recording calls for extensions that are contained within a huntgroup. We use record_session to record calls when made outbound or inbound to specific extensions, however we do this using the bridge function. Is it possible to record calls for a specific extension contained within a call group? For example can dial-string be defined with record_session? Has any body done this before? Many thanks Jon

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Under buffering if i use ICE SDP
by Alexander Haugg 20 Jul '17

20 Jul '17

Hi, i have the problem with underbuffering if i use ICE SDP. The good case: - ClientBria 4 - Codec G711 ulaw The bad case: - ICE Link API from Frozen Mountain - Codec G711 ulaw (no SRTP) I have two Wireshark traces. In the bad case i can see and hear the simulated silent parts every (round about) 500 ms. The bad case is independent of a codec. The same problem occured with G711 alaw and opus. Is it a known problem, or have anybody an idea? Is this a case for a Jira ticket? Thanks a lot!

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Adding video to audio call
by Gauri Kshirsagar 19 Jul '17

19 Jul '17

Hi, I am using Freeswitch version 1.9.0. I can make video calls. But when I try to add video to audio call it does not work. A makes audio call to B. Call is established. A adds video . ReINVITE sent to freeswitch has video added in SDP.Freeswitch sends 200 OK response for this INVITE to A which has video in SDP. But there is no ReINVITE being sent to B. I tried enabling renegotiate-codec-on-reinvite in vars.xml and also internal.xml Is this supported? If so is some other configuration required. Regards,

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Installed FreeSWITCH 1.6 / 1.9 on NetBSD
by Felipe Mateo 19 Jul '17

19 Jul '17

Hi, i have successfully installed freeswitch on NetBSD system, i am writing some initial documentation about this. -- *Felipe Mateo*. *Cel:* *+18294302505 <https://onlinefm.changeip.org/>* *Click for Webcall *

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Sharing information between an ESL client and Verto clients in a conference
by Mark Melling 19 Jul '17

19 Jul '17

Hi, I'd like to be able to share information between an Event Socket Library client and Verto clients in a conference. This may also include information that is shared before a Verto client has entered the conference, but they will still get access to this information. It looks like the way to achieve this is to use the Verto "live array" functionality. So on the assumption I'm correct how can an ESL client update the live array of a conference? And how can an ESL client receive any live array update events? Thanks Mark

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Verto member name
by 3b1den＠gmail.com 19 Jul '17

19 Jul '17

I`m trying to implement verto communicator into fusionpbx. But it doesnt show member name, only caller number on test vm. On working vm (fs builded from source 1.6.17) code looks like "<h4 class="chat-members-name" ng-class="{ 'clickable': verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)">" On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "<h4 class="chat-members-name clickable" ng-class="{ 'clickable': verto.data.confRole == 'moderator' }" ng-click="toggleModMenu($index)"> How to fix it? https://prnt.sc/fx1pqu

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dtmf_type under gateway
by Tihomir Culjaga 18 Jul '17

18 Jul '17

Hi, If i define a gateway in directory and have an external profile that scans gateways in directory... Is it possible to overwrite dtmf-type using variable dtmf_type under gateway definition for that sip profile? Example: sip_profile has set rfc2833 while the gw under that profile has the variable set to none (borh direction). Please, advice. Regards, Tihomir. Sent from my iPhone

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Verto member name
by 3b1den＠gmail.com 18 Jul '17

18 Jul '17

I`m trying to implement verto communicator into fusionpbx. But it doesnt show member name, only caller number on test vm. On working vm (fs builded from source 1.6.17) code looks like "[ALERT] mod_verto.c:5274 EVENT BROADCAST conference-liveArray.selector@domain { "data": { "action": "add", "arrIndex": 0, "name": "selector", "hashKey": "5cda1538-c6f2-f05b-2e3a-d92be05a2834", "wireSerno": 1, "data": ["0721", "admin@admin", "admin", "opus@48000", "{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,\"talking\":false,\"floor\":false,\"energyScore\":0},\"video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE VIDEO (BLIND)\"}", { "avatar": " http://gravatar.com/avatar/d41d8cd98f00b204e9800998ecf8427e.png?s=600" }, null] }, "eventChannel": "conference-liveArray.selector@domain" } On test vm (fusionpbx autoinstaller debian jessie 1.6.19) "[ALERT] mod_verto.c:5322 EVENT BROADCAST conference-liveArray.selector@domain { "data": { "action": "add", "arrIndex": 0, "name": "selector", "hashKey": "807bde07-ed9d-673e-1fa1-6ad24379916b", "wireSerno": 1, "data": ["0006", "1808", "1808", "opus@48000", "{\"audio\":{\"muted\":true,\"deaf\":false,\"onHold\":false,\"talking\":false,\"floor\":false,\"energyScore\":0},\"video\":{\"visible\":false,\"videoOnly\":false,\"avatarPresented\":false,\"mediaFlow\":\"sendOnly\",\"muted\":true,\"floor\":false,\"reservationID\":null,\"videoLayerID\":-1},\"oldStatus\":\"MUTE VIDEO (BLIND)\"}", { "email": "admin@admin", "avatar": " http://gravatar.com/avatar/a3175a452c7a8fea80c62a198a40f6c9.png?s=600" }, null] }, "eventChannel": "conference-liveArray.selector@domain" } How to fix it?

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secondary outbound gateway - how to prefix?
by jungle boogie 18 Jul '17

18 Jul '17

Hi All, I would like to have a secondary gateway added to freeswitch. To have it used for outbound calling, I need to prefix the dialstring with something like a 9 followed by the 10 digit number. I think that's what this page is explaining: https://freeswitch.org/confluence/display/FREESWITCH/Configuring+FreeSWITCH <condition field="destination_number" expression="^9(1\d{10})$"> Will that use the gateway if the telephone number is prefixed with a 9? Thanks!

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free testing tool - automated calling (SIP and/or webRTC)?
by robert mundkowsky 18 Jul '17

18 Jul '17

Anyone have suggestions on free automated calling toolsfor testing a system using SIP and/or webRTC? Robert

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