December 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

bridge_generate_comfort_noise not honoring integer value for volume level
by Matthew Meek 19 Dec '17

19 Dec '17

I can successfully inject whitenoise on the bridged leg (audio from A leg towards the bridged call) in the dial plan by setting bridge_generate_comfort_noise=true. The whitenoise is quite loud and I would like to make it quieter or silent (I am just trying to keep the far end firewall port open during long periods of A leg not sending packets due to hold). I have tried values of -1, 40,1400, and 4000 instead of true and they produce the same amount of whitenoise volume. The source code in switch_ivr_bridge.c looks like it indeed honors an integer value, but somehow the end result is always the same. If I do not set the variable I get no whitenoise, so the variable is indeed controlling whitenoise, just not the volume level of whitenoise. The codec is PCMU. I can repeat this with 1.2, 1.4, and 1.6 branches of FS. Matthew Meek

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Help me to install and run
by Ali Haider 19 Dec '17

19 Dec '17

Hi Everyone Hiiii I need help anyone help me how to install and run FreeSWITCH in Linux 16.04 Lts as well as Windows On windows I have already done steps till Visual studio soln when I build On visual studio 2013 it shows 182 error I can’t under stand how to remove it it is file missing error but I don’t know how to add files in git Sent from my iPhone

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how to block requests with From Ip equal to server interface IP?
by Miguel Jesús López Valverde 18 Dec '17

18 Dec '17

Good afternoon everyone I get a new query regarding a type of attack that our freeswitch servers receive constantly in case someone knows how to block them. These are INVITE or REGISTER requests in which the FROM: field arrives with the ip and port equal to the public interface of the server, so the different protection options that I have tried have not blocked these requests: - IpTables can not filter by the information From the INVITE message. - Fail2Ban is equally limited than IpTables. - ACLs have not resolved to filter these requests. Does anyone know any way to block these requests? I send here a trace with an INVITE message where you can see a request of this type. Thanks and best regards. U 2017/12/14 18:32:55.156886 185.107.94.121:11120 -> 182.30.1.194:5060 INVITE sip:390239297988@ 182.30.1.194:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 122.221.117.131:5060;branch=z9hG4bK-524287-1---xi3qy2kz737ni404. Max-Forwards: 70. Contact: <sip:15714000000@122.221.117.131:5060;transport=UDP>. To: <sip:390239297988@ 182.30.1.194;transport=UDP>. From: <sip:15714000000@ 182.30.1.194;transport=UDP>;tag=hlzg2jcv. Call-ID: KaQqH51mAcFv34qN8cGyv3... CSeq: 1 INVITE. Content-Type: application/sdp. User-Agent: Z 3.14.38765 rv2.8.3. Allow-Events: presence, kpml, talk. Content-Length: 0. . --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 138, Issue39
by allen＠praecom.com 17 Dec '17

17 Dec '17

You will probably have to set up media resources (medpro / media server) as well. Sent from my HTC ----- Reply message ----- From: freeswitch-users-request(a)lists.freeswitch.org To: <freeswitch-users(a)lists.freeswitch.org> Subject: FreeSWITCH-users Digest, Vol 138, Issue39 Date: Wed, Dec 13, 2017 12:12 PM Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: [EXTERNAL] Re: FS connecting to Avaya (Karanath sachidanandan, Sanooj) ---------------------------------------------------------------------- Message: 1 Date: Wed, 13 Dec 2017 18:11:49 +0000 From: "Karanath sachidanandan, Sanooj" <SANOOJ_KARANATH_SACHIDANANDAN(a)homedepot.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] [EXTERNAL] Re: FS connecting to Avaya Message-ID: <77695D16-34C1-4D3F-8946-6363C0F56820(a)homedepot.com> Content-Type: text/plain; charset="utf-8" Thanks Tom for pointers , we are looking more to peer with CM . Will look more into the direction you suggest . AES option was to connect to CM using JTAPI , but couldn’t make much progress in that dirction Regards Sanooj From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> on behalf of Tom Lynn <tom(a)tomlynn.com> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: Sunday, December 10, 2017 at 7:32 PM To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [EXTERNAL] Re: [Freeswitch-users] FS connecting to Avaya Are you planning to peer directly to Communication Manager or using Avaya Session Manager in between? It can be done both ways. If the requirement is to transact only with Communication Manager a direct connection may work out. In that case I'd suggest you'll need to add a node-name (which is like a host table entry) for FS and a signaling group of type SIP, along with a trunk group whose channels are controlled by said signaling group. Assume TCP signaling to begin with and pay attention to your port numbers. Some may already be used by other signaling groups. Codecs are assigned based on a network-region construct which is used to model your voip network against your physical topology, supporting call admission control and where DSP channels are sourced. Since Avaya doesn't use it's server host to mix audio, you'll have to know that the network region you choose either has DSPs or a logical path to a region which does is available. The network region is assigned to the SIP signaling group in a field called Far End Network region. To point calls at the trunk group you'll have to set up a route pattern and then place a dial string match in either an ARS analysis table (PSTN) or an AAR analysis table (private network). How are you thinking about using AES? I don't see an easy interface, but if you program TSAPI, it may be useful to get call events from the Avaya side. Tom Lynn On Fri, Dec 8, 2017 at 2:12 PM, Karanath sachidanandan, Sanooj <SANOOJ_KARANATH_SACHIDANANDAN(a)homedepot.com<mailto:SANOOJ_KARANATH_SACHIDANANDAN@homedepot.com>> wrote: Hi All, Is there any documentation on connecting FS to AVAYA CM/AES ? or any pointers ? Regards Sanooj ________________________________ The information in this Internet Email is confidential and may be legally privileged. It is intended solely for the addressee. Access to this Email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this Email are subject to the terms and conditions expressed in any applicable governing The Home Depot terms of business or client engagement letter. 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It is intended solely for the addressee. Access to this Email by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. When addressed to our clients any opinions or advice contained in this Email are subject to the terms and conditions expressed in any applicable governing The Home Depot terms of business or client engagement letter. The Home Depot disclaims all responsibility and liability for the accuracy and content of this attachment and for any damages or losses arising from any inaccuracies, errors, viruses, e.g., worms, trojan horses, etc., or other items of a destructive nature, which may be contained in this attachment and shall not be liable for direct, indirect, consequential or special damages in connection with this e-mail message or its attachment.

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Problem with personalized locutions and use of say with type shor_date_time.
by Miguel Jesús López Valverde 15 Dec '17

15 Dec '17

Hello Freeswitch users: I send this email to you with a new problem found. I have a set of complete locutions created with a TTS for Spanish locutions installed in Freeswitch, along with those brought by default in English. When using the phrases in Spanish, I am only having problems when using the directive: <action function = "say" data = "$ 1" method = "pronounced" type = "short_date_time" /> but with other directives "say" with different methods, I'm not having any problems. For example, when accessing voicemail using these locutions set in Spanish, I get the following error that prevents access to recorded messages: 2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (es_CB:es) 2017-12-13 16:58:52.252852 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 16:58:53.172924 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-new.wav 2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (es_CB:es) 2017-12-13 16:58:53.292911 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 16:58:54.832899 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/voicemail/vm-message_number.w av 2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (es_CB:es) 2017-12-13 16:58:54.932903 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 16:58:55.872903 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/es/colabora/sonia/digits/1.wav 2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [es_CB] 2017-12-13 16:58:55.972959 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (es_CB:es) 2017-12-13 16:58:55.972959 [ERR] mod_say_es.c:471 Unknown Say type=[18] By changing this language by default English, it works correctly ... 2017-12-13 17:18:05.327117 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-new.wav] (en:en) 2017-12-13 17:18:05.367151 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 17:18:05.667117 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-new.wav 2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-message_number.wav] (en:en) 2017-12-13 17:18:05.767136 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 17:18:06.587140 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-message_number.wav 2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1] (en:en) 2017-12-13 17:18:06.707138 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 17:18:07.407128 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://digits/1.wav 2017-12-13 17:18:07.507133 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:250 Handle say:[1513180033] (en:en) 2017-12-13 17:18:07.527108 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-12-13 17:18:10.427138 [DEBUG] switch_ivr_play_say.c:1942 done playing file file_string://time/today.wav!time/at.wav!digits/3.wav!digits/40.wav!digits/7 .wav!time/p-m.wav 2017-12-13 17:18:10.547147 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms It is proven that I have all the locutions in Spanish, including those in the "time" and "digits" folders. Does anyone know how I can get a proper voicemail operation with personalized locutions when using the function "say" with "type =" short_date_time "" ?. Thank you very much and best regards. --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus

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Re Event Priority setting and effect?
by Andrew Keil 15 Dec '17

15 Dec '17

To FreeSWITCH Users, I noticed the following inside FreeSWITCH Confluence: setPriority setPriority([$number]) Sets the priority of an event to $number in case it's fired. I assume this matches the Lua function: event:setPriority(x) >From switch_types.h: typedef enum { SWITCH_PRIORITY_NORMAL, SWITCH_PRIORITY_LOW, SWITCH_PRIORITY_HIGH } switch_priority_t; I have a couple of questions: If I have setup the remote end of an ESL connection to FreeSWITCH to listen for the following: event plain CUSTOM MY-EVENT log 5 event plain BACKGROUND_JOB 1. If I fire a CUSTOM event (MY-EVENT) from a Lua script on FreeSWITCH does using event:setPriority(x) have any effect on packet ordering received at the remote ESL end (ie. If I set the Priority to 2 (for SWITCH_PRIORITY_HIGH) will the CUSTOM event come before any "log" events or "BACKGROUND_JOB" events)? 2. Can I confirm that the only options available to my Lua script using event:setPriority is 0 (for SWITCH_PRIORITY_NORMAL (which I assume is the default)), 1 (for SWITCH_PRIORITY_LOW) and 2 (for SWITCH_PRIORITY_HIGH)? Thanks in advance, Andrew Keil

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FS or JSSIP/SIP.JS bug?
by Igor Olhovskiy 14 Dec '17

14 Dec '17

Hi! Run into a bug, when using «#» (hash) symbol in numbers using sip-over-wss with jssip or sip.js. Idea is, when you call some number, that contains # symbol, Freeswitch in a case of 2nd leg answer sends 200 OK with full dialed number in Contact field. And when jssip or sipjs tries to parse Contact field with hash symbol, they can’t do it (maybe cause this symbol is not urlencoded) So, is this a FS bug or more to JS sip libraries? Regards, Igor

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about sip hold and H264
by Madovsky 14 Dec '17

14 Dec '17

Dunno if it's a problem of configuration or an issue: if I hold the a-leg with a call with video H264 and unhold audio is ok but video received from the B-leg freezes. I have to hold the b-leg and unhold to let the a-leg continue to receive the b-leg video frames. is there any special configuration to solve it or I report a jira Thanks Franck

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Random number in variable
by Gregor Nanger 14 Dec '17

14 Dec '17

Hi, Guys! One hint, please. Is there a way to get random integer in FS variable without lua in Dialplan like: <action application="set" data=" myvar= some function to get random number betwee 100-200 or curl fetch? "/> Thank you, Gregor

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FS not reacting to BYE
by Igor Olhovskiy 14 Dec '17

14 Dec '17

Hi all! Got into issue with one of providers, and can’t get some things... For what reason FS is not reacting to BYE on external profile? FS -> Prov INVITE sip:3809XXXXXXXX@sip.telecomax.net SIP/2.0 Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKt7cUUFUKyea3r Max-Forwards: 69 From: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F To: <sip:3809XXXXXXXX@sip.telecomax.net> Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 116249601 INVITE Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 270 X-Serialnumber: 000413719560 P-Key-Flags: resolution="31x13", keys="4" X-accountcode: fusion.contactise.com X-FS-Support: update_display,send_info Remote-Party-ID: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1513140493 1513140494 IN IP4 138.68.65.164 s=FreeSWITCH c=IN IP4 138.68.65.164 t=0 0 m=audio 30838 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 FS <- Prov SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080 From: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F To: <sip:3809XXXXXXXX@sip.telecomax.net>;tag=as5cec60a1 Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 116249602 INVITE User-Agent: Free World Web Client Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:3809XXXXXXXX@80.242.134.195> Content-Type: application/sdp Content-Length: 268 v=0 o=root 21659 21659 IN IP4 80.242.134.195 s=session c=IN IP4 80.242.134.195 t=0 0 m=audio 13494 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv FS <- Prov SIP/2.0 200 OK Via: SIP/2.0/UDP 138.68.65.164:5080;branch=z9hG4bKUg6KXacQUQ0Nm;received=138.68.65.164;rport=5080 From: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F To: <sip:3809XXXXXXXX@sip.telecomax.net>;tag=as5cec60a1 Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 116249602 INVITE User-Agent: Free World Web Client Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:3809XXXXXXXX@80.242.134.195> Content-Type: application/sdp Content-Length: 268 v=0 o=root 21659 21660 IN IP4 80.242.134.195 s=session c=IN IP4 80.242.134.195 t=0 0 m=audio 13494 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv FS -> Prov ACK sip:3809XXXXXXXX@80.242.134.195 SIP/2.0 Via: SIP/2.0/UDP 138.68.65.164:5080;rport;branch=z9hG4bKvSZcZ5vtr0p8F Max-Forwards: 70 From: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F To: <sip:3809XXXXXXXX@sip.telecomax.net>;tag=as5cec60a1 Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 116249602 ACK Contact: <sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c> Proxy-Authorization: Digest username="456XXXXXXX", realm="Free World Web Client", nonce="6d32caff", algorithm=MD5, uri="sip:3809XXXXXXXX@sip.telecomax.net", response="c0aa24f024fc447594290b580bcedbfa" Content-Length: 0 <- This BYE is received, but no react to it in CLI at all. FS <- Prov BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0 Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport From: <sip:3809XXXXXXXX@sip.telecomax.net>;tag=as5cec60a1 To: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 102 BYE User-Agent: Free World Web Client Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 I've set up sofia debug all 9 and received this ------------------------------------------------------------------------ BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0 Via: SIP/2.0/UDP 80.242.134.195:5060;branch=z9hG4bK06dce488;rport From: <sip:3809XXXXXXXX@sip.telecomax.net>;tag=as77b4319c To: "456XXXXXXX" <sip:456XXXXXXX@sip.telecomax.net>;tag=gFjZSmajjg36F Call-ID: 7825a584-5aab-1236-22a3-525feb2d3359 CSeq: 102 BYE User-Agent: Free World Web Client Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 ------------------------------------------------------------------------ tport.c:3023 tport_deliver() tport_deliver(0x7ff1e0004650): msg 0x7ff1e0025700 (536 bytes) from udp/80.242.134.195:5080/sip next=(nil) nta.c:2880 agent_recv_request() nta: received BYE sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080;transport=udp;gw=3b179ac8-000f-453a-8059-7922501ae04c SIP/2.0 (CSeq 102) nta.c:3248 agent_aliases() nta: canonizing sip:gw+3b179ac8-000f-453a-8059-7922501ae04c@138.68.65.164:5080 with contact nta.c:3060 agent_recv_request() nta: BYE (102) going to existing leg nua_server.c:102 nua_stack_process_request() nua: nua_stack_process_request: entering nua_stack.c:271 nua_stack_event() nua(0x7ff204000b60): event i_bye 100 Trying nua_stack.c:359 nua_application_event() nua: nua_application_event: entering nta.c:7134 _nta_incoming_timer() nta: timer I fired, terminate 200 response nta.c:5825 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x7ff20dc4ac60) nta.c:7188 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free nta.c:1289 agent_timer() nta: timer not set tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e8004410): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e8004410) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e8004410) msg 0x7ff1e80c90c0 from (udp/138.68.65.164:5060) has 4 bytes, veclen = 1 tport.c:3023 tport_deliver() tport_deliver(0x7ff1e8004410): bad msg 0x7ff1e80c90c0 (4 bytes) from udp/176.104.56.91:5060/sip next=(nil) tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7ff1e0004650): events IN tport.c:2864 tport_recv_event() tport_recv_event(0x7ff1e0004650) tport.c:3205 tport_recv_iovec() tport_recv_iovec(0x7ff1e0004650) msg 0x7ff1e00582d0 from (udp/138.68.65.164:5080) has 536 bytes, veclen = 1 And this is it…. Also there is one-way sound, but I think it’s somehow related to this. FS 1.6.19 (FusionPBX) Regards, Igor

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