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conference member play image file
by William Simon 16 May '19

16 May '19

Using Freeswitch 1.8.5 release version we are trying to play a PNG image file to individual video conference members. Per confluence the API is: conference <confname> play [{vol=<volume>,full-screen=true,png_ms=100}]<file-path> [async|<member_id> [nomux]] Doing this API and omitting the details after the file-path, the PNG image is displayed to all conference members. If we include the member_id afterward in order to send the image to just one member, suddenly Freeswitch complains: 2019-05-10 10:23:11.288415 [ERR] mod_png.c:120 Channel has no video and audio_file is not set! To confirm, the participant is a video channel. “The information transmitted is intended only for the person or entity to which it is addressed and may contain proprietary, business-confidential and/or privileged material. If you are not the intended recipient of this message you are hereby notified that any use, review, retransmission, dissemination, distribution, reproduction or any action taken in reliance upon this message is prohibited. If you received this in error, please contact the sender and delete the material from any computer.”

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Conference duration limit
by Ramesh Kandasamy 16 May '19

16 May '19

Hi all, Is there a config to limit the duration of a conference from the time it is created? I came across sched_hangup. But if i configure this on dialplan it seems to be applicable per participant joining this conference (through this dialplan) but not for this conference. Thanks Ramesh

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Questions about FS
by Mohamed Saouli 16 May '19

16 May '19

Dear Madam, Mr, I am actually trying to implement your solution FreeSwitch for my company. And i have some requirements and questions, which are: - Can I configure rules to manipulate the sigalling on input and on output of FS? - To cope with NAT traversal issues, must I configure FS with a systematic relay of the media streams or is it possible to configure FS so that it relays the media streams only when it's necessary? Best regards, -- <https://audio.digigram.com/digigram-iqoya-range-the-most-advanced-aoip-code…>

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SRV and TTL
by David Villasmil 15 May '19

15 May '19

Hello all, I’m testing SRV where TTL is very low, like ~5 seconds. But I don’t see FS refreshing the SRV records. Is there a minimum TTL FS will honor? Thanks all David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Occasional Dropped call at 30 seconds
by Sean Devoy 15 May '19

15 May '19

Hi all, I have a problem with one client where the call is fine and they can talk for almost exactly 30 seconds, then the log says originating phone (of FS) hung up. This only happens like 1% to 5% of the time. The phones are Mitel 6867i and 6865i which must use TCP not UDP. The firewall is a $#!#&* SonicWall. I am 99% convinced it is the firewall. It happened once on a Cisco phone using UDP so I increased the firewall UDP timeout. I already have keep-alive set to 30 seconds. It has not recurred on UDP phones since. Any ideas on something to do to Mitel config or SonicWall that might prevent a drop after 30 seconds? I have considered PCAP, but with so few occurrences I am worried about PCAP size before I get a "hit." Thanks, Sean

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Verto conference: video garbage
by Xiemin Chen 15 May '19

15 May '19

There's video garbage showing something during the Verto video conference. Even though I enlarge the jitter buffer the problem still exists. Anyway to improve this? Sincerely Xiemin

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UNALLOCATED EXTENSION
by Sean Devoy 14 May '19

14 May '19

Hi, I have an internal extension (AASTRA/MITEL6867i phone) that is registered on the server and shows in "show registrations". However, when I attempt <action application="bridge" data="sofia/internal/110%somedom.bizfocused.com" /> I get [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 3770a2a3 2019-05-14 17:38:38.202824 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/110) State Change CS_ROUTING -> CS_CONSUME_MEDIA 3770a2a3 2019-05-14 17:38:38.202824 [DEBUG] switch_core_state_machine.c:643 (sofia/internal/110) State ROUTING going to sleep 3770a2a3 2019-05-14 17:38:38.202824 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/110) Running State Change CS_CONSUME_MEDIA (Cur 4 Tot 3889) 3770a2a3 2019-05-14 17:38:38.202824 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/110) State CONSUME_MEDIA 3770a2a3 2019-05-14 17:38:38.202824 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/110) State CONSUME_MEDIA going to sleep 3770a2a3 2019-05-14 17:38:38.242809 [DEBUG] sofia.c:7291 Channel sofia/internal/110 entering state [terminated][404] 3770a2a3 2019-05-14 17:38:38.242809 [NOTICE] sofia.c:8484 Hangup sofia/internal/110 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 3770a2a3 2019-05-14 17:38:38.242809 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/110) Running State Change CS_HANGUP (Cur 4 Tot 3889) 3770a2a3 2019-05-14 17:38:38.242809 [DEBUG] switch_core_state_machine.c:847 (sofia/internal/110) Callstate Change DOWN -> HANGUP Any ideas? Thanks, Sean

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codec error on 1.6
by Sandro Bordacchini 14 May '19

14 May '19

Hi everyone. I am facing a weird issue on a production setup, FS version 1.6.16~33~e6d643b-1~jessie+1 installed with deb from offcial repo, o.s. debian 8. Scenario is pretty straightforward: call incoming from voip provider and routed to a sip phone, i have no nat between FS and provider, i have nat between FS and sip phone. Provider sends INVITE with sdp (g711a + g729 + te) to FS FS sends INVITE with sdp (g711a + g729 + te) to sip phone sip phone sends 180 RINGING back to FS FS sends 183 SESS PROG with sdp (g711a + te) back to provider PRACK and 200OK between provider and FS sip phone sends 200 OK with sdp (g729 + te) to FS FS sends ACK to sip phone FS sends 200 without sdp to provider FS sends UPDATE with sdp (g729 + te + g711a) provider sends ACK to FS sip phone sends 200OK (to the UPDATE) with sdp (G729 + te) to FS At this point I have a bidirectional g711a rtp stream on the external profile. Instead on the internal profile i receive g729 from the phone but FS does not send any G729 stream to the phone. I have run some tests: if i don't have G711a codec as 1st choice in the incoming INVITE, the audio is ok; both the provider and the phone support both G711a and G729, but sip phone always answer ONLY with its preferred codec (in this test, G729). Generally speaking, if sip phone's preferred codec is different from the first codec in the incoming INVITE from the provider, i have the issue. Since the codec mix and order from the provider is not known, i cannot configure the sip phone accordingly. On both external and internal profiles i have: *inbound-codec-negotiation = generous* *inbound-late-negotiation = true* **-codec-prefs= G729, PCMA* FS logfile presents an error (loglevel: INFO, privacy data screened with *): 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:27:59.997037 [NOTICE] switch_channel.c:1104 New Channel sofia/external/********(a)******.net [922bb07e-8dce-4416-9073-52db21a8a432] 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:27:59.997037 [INFO] mod_dialplan_xml.c:637 Processing ******* <*******>->************ in context public 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:28:00.037033 [NOTICE] switch_ivr.c:2172 Transfer sofia/external/ ********(a)******.net to XML[6000@ ************] 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:28:00.037033 [INFO] mod_dialplan_xml.c:637 Processing ******** < ******** >->6000 in context ************ [...] fb16ac80-17f5-451b-a8ee-d22a135afc51 2019-05-09 17:28:00.197034 [NOTICE] switch_channel.c:1104 New Channel sofia/internal-bs/6000@192.168.50.119:5064 [fb16ac80-17f5-451b-a8ee-d22a135afc51] [...] 2019-05-09 17:28:00.277039 [INFO] sofia.c:1279 sofia/internal-bs/ 6000@192.168.50.119:5064 Update Callee ID to "6000 6000" <6000> fb16ac80-17f5-451b-a8ee-d22a135afc51 2019-05-09 17:28:00.277039 [NOTICE] sofia.c:7156 Ring-Ready sofia/internal-bs/6000@192.168.50.119:5064! 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:28:00.277039 [INFO] switch_ivr_originate.c:1215 Sending early media 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:28:00.277039 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/********(a)******.net! [...] fb16ac80-17f5-451b-a8ee-d22a135afc51 2019-05-09 17:28:03.497034 [NOTICE] sofia.c:8167 Channel [sofia/internal-bs/6000@192.168.50.119:5064] has been answered 922bb07e-8dce-4416-9073-52db21a8a432 2019-05-09 17:28:03.497034 [NOTICE] switch_ivr_originate.c:3632 Channel [sofia/external/********(a)******.net] has been answered fb16ac80-17f5-451b-a8ee-d22a135afc51 2019-05-09 17:28:03.577036 [ERR] sofia.c:8020 Codec Error! v=0 fb16ac80-17f5-451b-a8ee-d22a135afc51 o=- 1027803 1027804 IN IP4 192.168.50.119 fb16ac80-17f5-451b-a8ee-d22a135afc51 s=- fb16ac80-17f5-451b-a8ee-d22a135afc51 c=IN IP4 192.168.50.119 fb16ac80-17f5-451b-a8ee-d22a135afc51 t=0 0 fb16ac80-17f5-451b-a8ee-d22a135afc51 m=audio 25546 RTP/AVP 18 101 fb16ac80-17f5-451b-a8ee-d22a135afc51 a=rtpmap:18 G729/8000 fb16ac80-17f5-451b-a8ee-d22a135afc51 a=fmtp:18 annexb=no fb16ac80-17f5-451b-a8ee-d22a135afc51 a=rtpmap:101 telephone-event/8000 fb16ac80-17f5-451b-a8ee-d22a135afc51 a=fmtp:101 0-15 fb16ac80-17f5-451b-a8ee-d22a135afc51 a=ptime:20 fb16ac80-17f5-451b-a8ee-d22a135afc51 Matching packets time and logfile time, the "sofia.c:8020 Codec Error!" matches the sip phone 200OK answer to the UPDATE message. I tried to navigate the sofia.c code but i can't actually figure out what's happening. Please note that on a FS 1.4.9 setup (with a, i would say, "identical" configuration) this works, i.e. a bidirectional g729 stream is estabilished on the FS<->phone leg, also with identical signalling. I am wondering if there is any config parameter that should be edited/added/removed on the 1.6.x compared to the 1.4.x configuration. Btw, G729 commercial codec is active and running on both systems. Any hint is appreciated and, if needed i can post more data/log/tests. Thanks in advance. SB

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How to let freeswitch make a transfer call without dialplan xml?
by Leo Wolf 14 May '19

14 May '19

Hi, I want to make a transfer call on freeswitch. I use event socket to receive calls, do the calling invite logic in the program. Now I want to do the same things for transfering call. But the commands I found for transfer are all depending on dialplan xml. E.x., uuid_transfer, transfer, ..., etc. But what I want is do the transfer call without any xml, even the default. Is there any way to achieve that goal on freeswitch? Sincerely, Leo

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Failed Connection
by Sean Devoy 13 May '19

13 May '19

Hi, My user said "It just never connected" referring to an outbound call. It looks like everything is going great until the RTP is starting and then splat. This all happens in 2 seconds or less. I have tried different gateways and 2 different NAT'ed phone locations. The server is not NAT'ed. These have only started since I updated to FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit). But other things have changed too! I created a paste bin of my log: https://pastebin.com/9nw39DaR The lines in the log that confuse me are bordered by rows of ******. >From my clueless reviews, it looks like it may be related top CODEC (re)negotiation. Thanks for any insights you can share. Sean

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