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Re: [Freeswitch-users] VoIP encryption recommendations
by Joel Serrano 20 Feb '19

20 Feb '19

Bipin, Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client (requires linphone obviously), they also provide a server which receives such connections and pass over the unencrypted SIP+RTP to the backend. I think it's worth the try... some years ago it got around most blocks we tested. We ended up not implementing it but the initial tests did look good, don't know nowadays though... On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Bipin Patel <bipin(a)xbipin.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 23:58:21 +0400 > Subject: Re: [Freeswitch-users] VoIP encryption recommendations > > Btw we have tried almost all codecs and currently using g711u but it's > just impossible to get through easily not to mention the isp even > blacklists whole data center subnets if they find any data center to be a > safe heaven for VoIP providers. > > They even have this mechanism that home users if attempt to connect to any > blocked service then their IP ends up on a blacklist and then the filtering > gets worse for them until they reboot the router which gets them a new IP > and things get back to normal > > No wonder this region is considered a million dollar market where > Microsoft and Facebook itself can't manage to keep Skype and WhatsApp calls > even running no matter what. > > > On February 19, 2019 11:50:45 PM Bipin Patel <bipin(a)xbipin.com> wrote: > >> Hi, >> >> >> The whole sip protocol is blocked and udp VPN don't connect and tcp ones >> they delay packets a lot so calls end up heavily choppy. On mobile data the >> restrictions are even heavier and if packets are anywhere close to VoIP or >> VoIP over VPN etc they get filtered. Etisalat is the isp which buys >> blocking equipment from some vendor in UK who specialize in blocking VoIP >> and VPN. Last I was told by some person working there was they use a lot of >> L7 packet inspectors. >> >> Secondly it's not about setting up custom solutions for any company or >> client but we generate a lot of retail traffic so users need something that >> they can run on mobile etc like a customized dialer. Untill now I used to >> give them a openvpn profile which they used to run and then use Zoiper to >> place calls but all that is blocked now. >> >> Webrtc seems to work as of now coz it's new but there isn't a webrtc >> based mobile dialer till now which anyone can install and just use it to >> place calls >> >> >> >> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> In your case need >>> 1) increase ptime value to 40; >>> 2) use codec without compression; >>> 3) need to enable VAD feature; >>> >>> Also you can route torrent client network traffic via VPN together with >>> VoIP traffic. This will mask VoIP packets and not allow math your traffic >>> to VoIP profile on ISP equipment. >>> >>> >>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users < >>> freeswitch-users(a)lists.freeswitch.org>: >>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Bipin Patel <bipin(a)xbipin.com> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Bcc: >>>> Date: Tue, 19 Feb 2019 20:24:20 +0400 >>>> Subject: VoIP encryption recommendations >>>> hi, >>>> >>>> i would like the ask the community about VoIP encryption, currently in >>>> few middle east countries VoIP is officially blocked. The isp are so >>>> aggressive that they use all sorts of fancy tools to block it including >>>> skype calls, whatsapp calls etc and are very successful in doing it. So far >>>> companies like voipswitch and recently few others have been providing >>>> tunneling mechanisms to get over this but recently UDP traffic is heavily >>>> filtered and they go to the extreme of checking packet length and pattern >>>> and artificially introduce delay, jitter or simply block it if the number >>>> of hits are high. Switching to TLS/SRTP also doesnt help, it works with >>>> some isp but as soon as you try same using mobile data it stops working coz >>>> they match packet length and block based on the profile. ZRTP doesnt work >>>> coz a normal RTP streams needs to start and then it starts encrypting it >>>> but those initial RTP get blocked. >>>> >>>> With lack of any more VoIP encryption protocols its almost getting >>>> impossible to bypass block so has anyone have any ideas of any other modern >>>> form of encryption which can be used for VoIP (btw VPN are also blocked and >>>> more over if packet size increases then nothing works on mobile data). >>>> >>>> The market demand of skype replacements is also extremely high coz >>>> skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it >>>> and its blocked. >>>> >>>> -- >>>> Regards, >>>> Bipin >>>> >>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Bipin Patel via FreeSWITCH-users < >>>> freeswitch-users(a)lists.freeswitch.org> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Bcc: >>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST) >>>> Subject: [Freeswitch-users] VoIP encryption recommendations >>>> >>>> _________________________________________________________________________ >>>> 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 >>> >>> >> > > > > ---------- Forwarded message ---------- > From: Bipin Patel via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST) > Subject: Re: [Freeswitch-users] VoIP encryption recommendations > _________________________________________________________________________ > 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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[ERR] too many stacked extensions
by 梅表逵 20 Feb '19

20 Feb '19

Hi, I’m running freeswitch version 1.8.1. In the log files, I saw a lot of [ERR] too many stacked extensions. It happens when freeswitch is trying to bridge a callcenter member to agent. I don’t know how to handle it. Could you please give any advises? 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] switch_ivr_bridge.c:1063 sofia/internal/137095@10.20.86.100:5060 CUSTOM HOLD 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/137095@10.20.86.100:5060) State CONSUME_MEDIA going to sleep 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR] switch_ivr.c:902 Error sofia/internal/137095@10.20.86.100:5060 too many stacked extensions 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR] switch_ivr.c:902 Error sofia/internal/137095@10.20.86.100:5060 too many stacked extensions 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] switch_ivr_bridge.c:1744 (sofia/internal/137095@10.20.86.100:5060) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR] switch_ivr.c:902 Error sofia/internal/137095@10.20.86.100:5060 too many stacked extensions 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [ERR] switch_ivr.c:902 Error sofia/internal/137095@10.20.86.100:5060 too many stacked extensions 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/137095@10.20.86.100:5060) Running State Change CS_EXCHANGE_MEDIA (Cur 2 Tot 4919) 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/137095@10.20.86.100:5060) State EXCHANGE_MEDIA 55791bf8-30bb-11e9-a4fd-7125b8df8c7a 2019-02-15 08:47:56.419673 [DEBUG] mod_sofia.c:645 SOFIA EXCHANGE_MEDIA XML: <condition field="destination_number" expression="^(31\d{4})$" > <action application="set" data="dialed_extension=$1"/>  <action application="set" data="recording_follow_transfer=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="say_agent_first=1" /> <action application="set" data="bridge_pre_execute_aleg_app=lua"/> <action application="set" data="bridge_pre_execute_aleg_data=broadcastAgent.lua"/> <action application="set" data="call_type=q" /> <action application="bind_meta_app" data="4 ab s execute_extension::att_xfer XML features"/> <action application="callcenter" data="$1"/> </condition> </extension> Thanks Ceniy Mei

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switch_ivr
by Juan Nino 20 Feb '19

20 Feb '19

HI chekcing the logs I have seen actions taken by switch_ivr.c is there any way I can disable it? and/or check how does it work? I have commented mod_ivr but this does not seem to affect thanks

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Unsubscribe
by Ryan Hallett 19 Feb '19

19 Feb '19

Sent from Nine<http://www.9folders.com/>

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Re: [Freeswitch-users] VoIP encryption recommendations
by Sergey Safarov 19 Feb '19

19 Feb '19

In your case need 1) increase ptime value to 40; 2) use codec without compression; 3) need to enable VAD feature; Also you can route torrent client network traffic via VPN together with VoIP traffic. This will mask VoIP packets and not allow math your traffic to VoIP profile on ISP equipment. вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org>: > > > > ---------- Forwarded message ---------- > From: Bipin Patel <bipin(a)xbipin.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 20:24:20 +0400 > Subject: VoIP encryption recommendations > hi, > > i would like the ask the community about VoIP encryption, currently in few > middle east countries VoIP is officially blocked. The isp are so aggressive > that they use all sorts of fancy tools to block it including skype calls, > whatsapp calls etc and are very successful in doing it. So far companies > like voipswitch and recently few others have been providing tunneling > mechanisms to get over this but recently UDP traffic is heavily filtered > and they go to the extreme of checking packet length and pattern and > artificially introduce delay, jitter or simply block it if the number of > hits are high. Switching to TLS/SRTP also doesnt help, it works with some > isp but as soon as you try same using mobile data it stops working coz they > match packet length and block based on the profile. ZRTP doesnt work coz a > normal RTP streams needs to start and then it starts encrypting it but > those initial RTP get blocked. > > With lack of any more VoIP encryption protocols its almost getting > impossible to bypass block so has anyone have any ideas of any other modern > form of encryption which can be used for VoIP (btw VPN are also blocked and > more over if packet size increases then nothing works on mobile data). > > The market demand of skype replacements is also extremely high coz skype, > hangouts, whatsapp video, instagram video, viber etc etc, u name it and its > blocked. > > -- > Regards, > Bipin > > > > > > ---------- Forwarded message ---------- > From: Bipin Patel via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST) > Subject: [Freeswitch-users] VoIP encryption recommendations > _________________________________________________________________________ > 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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where g729 fs-latest-installer script now ?
by Yuriy Nasida 19 Feb '19

19 Feb '19

Hi guys, Does anybody know why site http://files.freeswitch.org/g729/ contain only install instrunction now ? Where fs-latest-installer-v1.6 script or something like this ? Please advice.

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Re: [Freeswitch-users] Opus code change
by Dragos Oancea 19 Feb '19

19 Feb '19

No, no changes in the library. On Tue, Feb 19, 2019 at 4:29 PM kaiduan xie via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kaiduan xie <kaiduanx(a)yahoo.ca> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 15:51:55 +0000 (UTC) > Subject: Opus code change > Hi, > > Did Freeswitch team make any source code changes to upstream Opus Opus > Codec <http://opus-codec.org/>? If yes, what are the changes? > > Thanks, > > /Kaiduan > > > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 08:29:20 -0800 (PST) > Subject: [Freeswitch-users] Opus code change > _________________________________________________________________________ > 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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VoIP encryption recommendations
by Bipin Patel 19 Feb '19

19 Feb '19

hi, i would like the ask the community about VoIP encryption, currently in few middle east countries VoIP is officially blocked. The isp are so aggressive that they use all sorts of fancy tools to block it including skype calls, whatsapp calls etc and are very successful in doing it. So far companies like voipswitch and recently few others have been providing tunneling mechanisms to get over this but recently UDP traffic is heavily filtered and they go to the extreme of checking packet length and pattern and artificially introduce delay, jitter or simply block it if the number of hits are high. Switching to TLS/SRTP also doesnt help, it works with some isp but as soon as you try same using mobile data it stops working coz they match packet length and block based on the profile. ZRTP doesnt work coz a normal RTP streams needs to start and then it starts encrypting it but those initial RTP get blocked. With lack of any more VoIP encryption protocols its almost getting impossible to bypass block so has anyone have any ideas of any other modern form of encryption which can be used for VoIP (btw VPN are also blocked and more over if packet size increases then nothing works on mobile data). The market demand of skype replacements is also extremely high coz skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it and its blocked. -- Regards, Bipin

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Opus code change
by kaiduan xie 19 Feb '19

19 Feb '19

Hi, Did Freeswitch team make any source code changes to upstream Opus Opus Codec? If yes, what are the changes? Thanks, /Kaiduan

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How to disable FS DTMF detection
by Adrian Worutowicz 19 Feb '19

19 Feb '19

Hi, I've developed a couple of endpoints and I use bgapi originate & bridge. Free Switch intercepts and interprets DTMF and I do not want that. In the dialplan I tried: <param name="pass-rfc2833" value="true"/> <action application="set" data="pass_rfc2833=true"/> <action application="set" data="exec_after_bridge_app=stop_dtmf"/> <action application="stop_dtmf"> <action application="spandsp_stop_dtmf"/> None of those methods works. Maybe there is a parameter in originate that I should use? *** How to tell Free Switch to leave alone the DTMF tones? *** I'll appreciate any help, Adrian.

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