November 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Failed DTMF sanity check
by Saqib Nawaz 21 Nov '19

21 Nov '19

Hi Guys, I'm getting this error message printed on my console [ERR] switch_rtp.c:2013 Failed DTMF sanity check. Why this is popping up ? how can I prevent this from occurring? Regards, Saqib Nawaz.

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Calling a destination and playing a pre-recorded message.
by Paul Mateer 19 Nov '19

19 Nov '19

Hi. I have a FreeSWITCH system up and running and I can make calls from one device to another which is fine. What I would now like to do is integrate it with another system so that when certain events occur in this other system, it dials one or more destinations and plays a pre-recorded message before hanging up. Can the playing of the sound file be hooked into a custom dialplan? I ask because a little experimentation has revealed that the dialplan is processed on the inbound channel (which seems entirely reasonable) which means that using playback in the dialplan results in audio sent to the caller, rather than the called. If this can't be achieved via the dialplan, does this mean that I need to create a specific client that can call a destination and somehow pipe sound into the input channel, rather than taking it from whatever input device might be available? Thanks for any assistance offered. Paul Sent from my Windows 10 device

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Private IP used for audio even though ACL should block it
by David P 18 Nov '19

18 Nov '19

In the log snippet below, a private IP is chosen for the audio connection (6th line from bottom) even though our acl.conf.xml should reject it. How can we be sure our ACL is enforced? Our version on debian9: FreeSWITCH Version 20.19.4-release-12-fc9d51c~64bit (-release-12-fc9d51c 64bit) Our autoload_configs/acl.conf.xml contains:  <list name="companyname-disallow-privateIPv4" default="allow"> <node type="deny" host="10.0.0.0" mask="255.255.255.0"/> <node type="deny" host="192.168.0.0" mask="255.255.0.0"/> </list> And our verto.conf.xml's "default-v4" profile references it: <param name="apply-candidate-acl" value="companyname-disallow-privateIPv4"/> (Also, strangely, we get a relay candidate from our TURN server for video but not audio.) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e m=audio 57658 UDP/TLS/RTP/SAVPF 111 103 9 102 0 8 105 13 110 113 126 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e c=IN IP4 10.0.0.90 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=rtcp:9 IN IP4 0.0.0.0 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2964267771 1 udp 2113937151 10.0.0.90 57658 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2509878665 1 udp 2113939711 2601:ipv6:address 57659 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e m=video 12769 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 127 125 104 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e c=IN IP4 52.public.turn.addr 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=rtcp:9 IN IP4 0.0.0.0 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2964267771 1 udp 2113937151 10.0.0.90 52903 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2509878665 1 udp 2113939711 2601:ipv6:address 52904 typ host generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e a=candidate:2563966249 1 udp 16785151 52.public.turn.addr 12769 typ relay raddr 73.user.public.ip rport 49605 generation 0 network-cost 999 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_codec.c:111 verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 Original read codec set to opus:116 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.0.90:57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4227 Drop audio Candidate cid: 1 proto: udp type: host addr: 2601:ipv6:address:57659 (no network path) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_vpx.c:703 VPX VER:v1.7.0 VPX_IMAGE_ABI_VERSION:4 VPX_CODEC_ABI_VERSION:8 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_vpx.c:703 VPX VER:v1.7.0 VPX_IMAGE_ABI_VERSION:4 VPX_CODEC_ABI_VERSION:8 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:3598 Set VIDEO Codec verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 VP8/90000 0 ms 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save video Candidate cid: 1 proto: udp type: host addr: 10.0.0.90:52903 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4227 Drop video Candidate cid: 1 proto: udp type: host addr: 2601:ipv6:address:52904 (no network path) 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4233 Save video Candidate cid: 1 proto: udp type: relay addr: 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtcp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4325 Look for Relay Candidates as last resort 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4278 Searching for rtp candidate. 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4287 Choose rtp candidate, index 1, 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4053 verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1 choosing family v4 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4298 Choose same candidate, index 0, for rtcp based on rtcp-mux attribute 52.public.turn.addr:12769 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4350 setting remote video ice addr to index 1 52.public.turn.addr:12769 based on candidate 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:4385 Setting remote rtcp video addr to 52.public.turn.addr:12769 based on candidate 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:8600 AUDIO RTP [verto.rtc/0300_db262ee9-b897-4ece-83f6-ca4507489bc9_v-1*c-1*f-1] 10.0.0.192 port 26664 -> 10.0.0.90 port 57658 codec: 111 ms: 20 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_rtp.c:4480 Starting timer [soft] 960 bytes per 20ms 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_core_media.c:8819 Activating RTCP PORT 57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [DEBUG] switch_rtp.c:4880 RTCP send rate is: 1000 and packet rate is: 20000 Remote Port: 57658 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [INFO] switch_rtp.c:3808 Activate RTP/RTCP audio DTLS client 8e2ab29f-cdc8-84d8-a804-cabb3d0a474e 2019-11-15 06:14:09.477153 [INFO] switch_rtp.c:3975 Changing audio DTLS state from OFF to HANDSHAKE

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Fwd: contact missing in 180 response
by Gaurang Gohil 18 Nov '19

18 Nov '19

---------- Forwarded message --------- From: Gaurang Gohil <gaurang.gohil(a)ecosmob.com> Date: Thu, Nov 7, 2019 at 6:40 PM Subject: contact missing in 180 response To: <mailman-owner(a)lists.freeswitch.org> Hello all, I set up a presence in linphone SDK and have to face the problem of contact loss in Sip Dialog cause of linephone SDK not add contact header in 180 or 183 packets, I tried to debug and found that in Sofia.c when we insert dialog in sip dialog table at the time of 180 or 183 with state of "early" but linephone does not add CONTACT header in 180 packet so Sofia unable to get contact header in 180 packet and due to this we lose the presence state when FS receive subscribe and unable to get contact in sip dialog table, please help me on this. please find the pcap i have attached to this email. -- Regards, *Gaurang Gohil* | Jr. Software Developer +91 9662008820 | Hangout: gaurang.gohil(a)ecosmob.com <jankit.shah(a)ecosmob.com> [image: Ecosmob Technologies Pvt. Ltd.] <https://htmlsig.com/t/000001C50ZHN> Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data ssdsds <https://twitter.com/ecosmob> sasadsdasdasdasdasdas <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- Regards, *Gaurang Gohil* | Jr. Software Developer +91 9662008820 | Hangout: gaurang.gohil(a)ecosmob.com <jankit.shah(a)ecosmob.com> [image: Ecosmob Technologies Pvt. Ltd.] <https://htmlsig.com/t/000001C50ZHN> Ecosmob Technologies Pvt. Ltd. https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data ssdsds <https://twitter.com/ecosmob> sasadsdasdasdasdasdas <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. Any opinion and other statement contained in this message and any attachment are solely those of the author and do not necessarily represent those of the company. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

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Unanswered pstn outbound call via gateway detection
by Md,Mehedi Hasan Kabir(Tanim) 18 Nov '19

18 Nov '19

Hi I have originated outbound call to pstn number using gateway provider flowroute with originate command. I am using python ESL to achieve this. If the call is not answered by pstn user, then i need to post status to my webserver . Can anyone suggests me how i can detect whether the pstn call is answered or not? Thanks Tanim

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Re: [Freeswitch-users] Performance questions
by Jose Fco. Irles Durá 16 Nov '19

16 Nov '19

> Jose, do you have any problem with current configurations? Some users reports audio problems: - some robotic calls - some calls, after a long time talking ok, listen robotic I'm finding possible causes, but the 95% of users that reports problems use Grandstream and I don't if the problem is mine or the phone. I also testing my own network infrastructure and I'm making heuristic captures for check quality. The machine with more load (2 sockets, 16 cores, 32 threads), has about 30% cpu usage at peak hours and 300k context switches. The normal cpu usage in peak hours has about 10% in the other servers Best regards -- Jose

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Performance questions
by Jose Fco. Irles Durá 16 Nov '19

16 Nov '19

Hi, I have several nodes with FreeSWICH, typically 2 socket Xeon v2 (with 8-12 cores every socket) and 64~128GB of RAM. I'm testing different configurations for max performance. I run FreeSWITCH containers (with docker and macvlan network), the base system is Ubuntu 18.04 and I run 4 containers for machine (max concurrency about 500-700 calls and max cps 200, limited in FreeSWITCH config). But I have some questions about the base config of my servers: 1. Is better cpu hyperthreading enabled or disabled? 2. Is relevant NUMA placement for FreeSWITCH (now I bind 2 FreeSWITCH processes to one NUMA node of the machine)? 3. Is it recommended cpu pinning to avoid excessive context switching? (my opinion is yes, but I don't know if the performance difference is too high) 4. I run default Ubuntu kernel (generic ubuntu flavor, with kernel CONFIG_HZ_250=y), but I can change to the "low latency" version, with CONFIG_HZ=1000. Witch is better? I have tested the two versions with timer_test command in FreeSWITCH and I haven't seen any differences (with no load): Avg: 20.000ms Total Time: 1000.002ms Any other advise about network tunning? PD: the containers are Debian Buster with FreeSWITCH 1.10.1 Best regards -- Jose

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Re: [Freeswitch-users] Using Curl in dial plan
by Faisal Hanif 16 Nov '19

16 Nov '19

Best option to use hangup_hook On Sat, 16 Nov 2019, 5:58 pm H Yavari via FreeSWITCH-users, < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: H Yavari <hyavari(a)rocketmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sat, 16 Nov 2019 12:34:55 +0000 (UTC) > Subject: Using Curl in dial plan > Hi to all, > > I am going to use curl and post some info like CID after call hang up to > another web server. Using the Lua and working with events is the only > solution or we have access to call variables in the dial plan when call is > finished or voicemail is done? > > Some useful hints would be appreciated. > > Regards. > H.Yavari > > > > ---------- Forwarded message ---------- > From: H Yavari via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org > > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sat, 16 Nov 2019 04:58:37 -0800 (PST) > Subject: [Freeswitch-users] Using Curl in dial plan > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com

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Using Curl in dial plan
by H Yavari 16 Nov '19

16 Nov '19

Hi to all, I am going to use curl and post some info like CID after call hang up to another web server. Using the Lua and working with events is the only solution or we have access to call variables in the dial plan when call is finished or voicemail is done? Some useful hints would be appreciated. Regards.H.Yavari

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Run Bash script in Dialplan
by Abel Monzon 15 Nov '19

15 Nov '19

Hello. I am trying to run a bash script when dialplan is generate. I have tested this bash script in CLI and is working as expected, but when is called from dialplan I don't see any error and looks like the bash was never runned. api = freeswitch.API(); rotation = api:executeString("system sh /opt/ASTPP/freeswitch/scripts/rotation.sh"); --> this rotate between 2x gateway name and save the name to a file, but looks like never run. gateway = api:executeString("system sh /opt/ASTPP/freeswitch/scripts/random2.sh"); ---> this script read the name of the gateway from the file, and works perfect. table.insert(xml, [[<action application="bridge" data="[]]..chan_var..[[]sofia/gateway/]]..*gateway* ..[[/]]..destination_number..[["/>]]); Here is the result on console: EXECUTE sofia/default/14254358039(a)198.50.167.93 bridge([leg_timeout=0,absolute_codec_string=^^:PCMA:PCMU:G729]sofia/gateway/GATEWAY-01/38227043) I don't any way to troubleshoot why 1x script run perfectly and the other one is not running. Thank You in advance. Abel

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