November 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Debian and Centos not officialy supported by server
by Federico Castro 16 Nov '18

16 Nov '18

Hi all, I installed FS on a HP DL-120 Gen9 server using Debian but reading server specifications I found that Debian and Centos are not officially supported for that server (and I realized that many others does not support too). Anyway I have everything installed and running except for RAID controller because there’s no drivers for that. So, trying to avoid unexpected behaviors on production I would like to know if any of you have some experience on using hardware that does not officialy support Debian/Centos? Thanks in advance.

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Since recently no ring back
by Paul Muaddib 16 Nov '18

16 Nov '18

Hello Freeswitch Users, Since recently I am not getting ring back singles for some phones that do inbound calls. Voice connection is working in both direction when the call is established. When someone calls where the ring back is not working then there is only a short beep and then silence until someone answers the phone. <action application="set" data="ringback=${de-ring}"/> <action application="ring_ready" /> <action application="bridge" data="user/18"/> Without bridge is is working. <action application="ring_ready" /> <action application="sleep" data="6000"/> <action application="answer"/> <action application="playback" data="$${hold_music}"/> Freeswitch sits behinde nat. Fresswitch Version 1.8.2 Best regards, Paul.

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RTP Update
by Vishal Dalsania 16 Nov '18

16 Nov '18

Is it possible to update RTP IP address after call is established without sending any SIP message?

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dynamic gateways
by Juan Pablo L 15 Nov '18

15 Nov '18

Hello, i m trying to load gateway information for profile dynamically, at the moment i am trying to do it in the directory section, when the profile is reload/rescan/restarted, upon directory request i m answering this xml: <document type="freeswitch/xml"> <section name="directory"> <users> <user id="TestCustomerPBX"> <gateways> <gateway name="TestCustomerPBX"> <param name="username" value="testpbx1"></param> <param name="password" value="abcd123"></param> <param name="proxy" value="X.X.X.X"></param> <param name="register" value="false"></param> <param name="caller-id-in-from" value="true"></param> </gateway> </gateways> </user> </users> </section> </document> but it is not working, i have tried like dozens of times with different parameters, values etc but none seems to work , even though the profile is reloaded/rescanned/restarted successfully the gateway(s) do not show in "sofia status" and when i attempt to make a call, naturally, i get an "invalid gateway" error, does anyone know what is the correct format of the xml to be answered ? ... other parameters i have tried are domain,profile,profiles and others alike and all possible combinations of them all .... and yes the gateway information is correct it works if loaded from xml file ... thanks!

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New FreeSWITCH Community Resources - SignalWire.community
by Anthony Minessale 15 Nov '18

15 Nov '18

Hi, I wanted to let everyone know that, as part of our effort to expand our community resources, we have started building out some resources. We are working on a forum, some github projects for FreeSWITCH and SignalWire tools and we started a slack space. I personally have resisted more modern tools, i've been on irc for decades (and we still are #freeswitch on freenode) but our community is growing and we have all gotten more mobile and using multiple devices and we want to make sure we make our community accessible to everyone. This mailing list will always be here, its running strong for 15 years now! Here is the link to the signalwire.community slack. Come join and lets discuss how we can continue to innovate in the communications industry. https://goo.gl/1mEBwn -- Anthony Minessale II Founder, FreeSWITCH. http://freeswitch.com https://youtu.be/l_hOxzCt6X4 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Problem with innovaphone devices
by Daniel 15 Nov '18

15 Nov '18

Dear all, I'm working with FS since a week. Hope, it's ok to use it in FusionPBX ;-) I found out many cool things so far but with one thing I'm stuck. I'm not able to initiate a connection from innovaphone devices (1 deskphone IP240 and 1 PBX/GW IP800). Calling to these innovaphones is ok. From other devices to FS and softphones I can connect too. In my first try from the innovaphones I got the message 2018-11-10 23:41:58.763840 [WARNING] switch_core_media.c:4181 NO candidate ACL defined, Defaulting to wan.auto 2018-11-10 23:41:58.763840 [NOTICE] sofia.c:7774 Hangup sofia/internal/11(a)pbx.[anonym].net [CS_NEW] [INCOMPATIBLE_DESTINATION] I found the solution to put apply-candidate-acl=localnet.auto to the internal SIP profile. I don't understand why this was necessary only for the innovaphones and not for other devices but after changing this I can establish a connection but cannot hear anything on both ends. After a few seconds the connection is disconnected automatically. Here's the log. I hope, someone can help https://pastebin.freeswitch.org/view/db088714 Best regards Daniel

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Hangup after bridge
by Mody, Darshan (Darshan) 13 Nov '18

13 Nov '18

Hi I find below documentation for hangup after bridge hangup_after_bridge BOOLEAN Controls what happens to a calling (A) party when in a bridge state and the called (B) party hangs up. If true the dialplan will stop processing and the A leg will be terminated when the B leg terminates. If false (default) the dialplan continues to be processed after the B leg terminates. This is checked after park_after_bridge<https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#…> and transfer_after_bridge<https://freeswitch.org/confluence/display/FREESWITCH/Variables+Master+List#…>. In the below code within switch_ivr_bridge.c we are checking park_after_bridge transfer after bridge (highlighted). However we are not checking for hangup_after_bridge setting and hanging up the channel. Is there any reason for missing the check of hangup_after_bridge flag before hanging up the channel? static switch_status_t audio_bridge_on_exchange_media(switch_core_session_t *session) { switch_channel_t *channel = switch_core_session_get_channel(session); switch_ivr_bridge_data_t *bd = switch_channel_get_private(channel, "_bridge_"); switch_channel_state_t state; const char *var; if (bd) { switch_channel_set_private(channel, "_bridge_", NULL); if (bd->session == session && *bd->b_uuid) { audio_bridge_thread(NULL, (void *) bd); switch_core_session_reset(session, SWITCH_TRUE, SWITCH_TRUE); } else { switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); } } else { switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER); } switch_channel_clear_state_handler(channel, &audio_bridge_peer_state_handlers); state = switch_channel_get_state(channel); if (!switch_channel_test_flag(channel, CF_TRANSFER) && !switch_channel_test_flag(channel, CF_REDIRECT) && !switch_channel_test_flag(channel, CF_XFER_ZOMBIE) && bd && !bd->clean_exit && state != CS_PARK && state != CS_ROUTING && state == CS_EXCHANGE_MEDIA && !switch_channel_test_flag(channel, CF_INNER_BRIDGE)) { if (state < CS_HANGUP && switch_true(switch_channel_get_variable(channel, SWITCH_PARK_AFTER_BRIDGE_VARIABLE))) { switch_ivr_park_session(session); return SWITCH_STATUS_FALSE; } else if (state < CS_HANGUP && (var = switch_channel_get_variable(channel, SWITCH_TRANSFER_AFTER_BRIDGE_VARIABLE))) { transfer_after_bridge(session, var); return SWITCH_STATUS_FALSE; } if (switch_channel_test_flag(channel, CF_INTERCEPTED)) { switch_channel_clear_flag(channel, CF_INTERCEPT); switch_channel_clear_flag(channel, CF_INTERCEPTED); return SWITCH_STATUS_FALSE; } else { if (switch_channel_test_flag(channel, CF_INTERCEPT)) { switch_channel_hangup(channel, SWITCH_CAUSE_PICKED_OFF); } else { if (!switch_channel_test_flag(channel, CF_ANSWERED)) { int x = 0; if (switch_channel_execute_on(channel, "execute_on_orphaned_bleg") == SWITCH_STATUS_SUCCESS) { x++; } if (switch_channel_api_on(channel, "api_on_orphaned_bleg") == SWITCH_STATUS_SUCCESS) { x++; } if (!x) { switch_channel_hangup(channel, SWITCH_CAUSE_ORIGINATOR_CANCEL); } } else { switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING); } } } } if (switch_channel_get_state(channel) == CS_EXCHANGE_MEDIA) { switch_channel_set_variable(channel, "park_timeout", "3"); switch_channel_set_state(channel, CS_PARK); } return SWITCH_STATUS_FALSE; } Thanks Darshan

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How to avoid transcoding?
by Ulrich Backes 13 Nov '18

13 Nov '18

Hi guys, I've investigated many hours and cannot solve this problem: A and B should use g711u. A: B: But FreeSwitch decides to transcode: The settings: vars.xml <X-PRE-PROCESS cmd="set" data="media_mix_inbound_outbound_codecs=false"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,G722,PCMU,PCMA,VP8"/> internal.xml <param name="inbound-codec-negotiation" value="greedy"/> <param name="inbound-late-negotiation" value="true"/> <param name="disable-transcoding" value="true"/> <param name="inbound-proxy-media" value="false"/> <param name="inbound-bypass-media" value="false"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> default.xml (Using the demo-Dialplan) <extension name="Local_Extension"> ... Using FS v 1.8 Thanks. Uli

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What is the recommended practice to use uuid_bridge?
by Mody, Darshan (Darshan) 13 Nov '18

13 Nov '18

Hi We have a FS which receives inbound calls (IB). Using FS's ESL we originate outbound calls (OB). We want to bridge the media of inbound to outbound calls. Should we be using uuid_bridge command with arguments IB and OB (uuid_bridge IB OB) or should we be using (uuid_bridge OB IB). Thanks Darshan

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Audio interfacing to POTS port for WebRTC
by Rick Jarvis 12 Nov '18

12 Nov '18

Strange request, so forgive me, but I’m looking to connect a PC running FS Verto/WebRTC to a telephone balancing unit. So basically, stereo audio in and out from a PC, to plug into an FXO port on a mixing console. The line switching on the console is academic as it will be permanently connected, and any dialling controlled from WebRTC. Appreciate this is a more of an electronics question, but if anyone knows a way to achieve this, someone on here will! Or maybe there’s a better way that I haven’t thought of!? Thanks R

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