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Problem with SRTP in HA
by Francisco Carmona Ramírez 05 Jul '19

05 Jul '19

Hi, i have some issues trying to make a HA with SRTP, i have that dialplan <extension name="salientes"> <condition field="destination_number" expression="^(\d{9})$"> <action application="set" data="ringback=${es-ring}"/> <action application="set" data="transfer_rinback=$${hold_music}"/> <action application="set" data="call_timeout=30" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="export" data="rtp_secure_media=optional"/> <action application="export" data="rtp_secure_media_outbound=forbidden"/> <action application="bridge" data="sofia/gateway/providerExt/$1"/> <action application="hangup"/> </condition> </extension> When i make a normal call in my FS1 it goes well, same when i do a normal call in a FS2, then the problem that i find is when i do a normal call in my FS1 (it takes AES_CM_128_HMAC_SHA1_32) and i do a "fsctl crash" in my "fs_cli", when the FS2 recovery the call, the audio drops in both ways, and if I use sngrep to see what is sending in the invite, i see that the crypto is "AES_CM_NULL_AUTH", in addition in the fs_cli on FS2 y found that error 2019-07-04 10:37:38.286877 [ERR] switch_core_media.c:1554 Error! No crypto to parse 2019-07-04 10:37:38.286877 [INFO] switch_rtp.c:4104 Activating audio Secure RTP SEND 2019-07-04 10:37:38.286877 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 2019-07-04 10:37:38.286877 [INFO] switch_rtp.c:4082 Activating audio Secure RTP RECV 2019-07-04 10:37:38.286877 [DEBUG] switch_core_sqldb.c:2617 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_32 Can someone give me some clarity about that problem? How I can solve this?

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FreeSWITCH Weekly Community Conference Call
by Ken Rice 03 Jul '19

03 Jul '19

Hey FreeSWITCHers. Join us today at 12noon US Central Time at https://conference.freeswitch.org/vc and call 888 or via sip:888@conference.freeswitch.org Open Floor discussion of FreeSWITCH and related technologies. This week we'll be talking about Debian 10's pending release! Plus more. Need help with FreeSWITCH? Come on in and ask your question. K

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Get videocall image preview in target endpoint
by Jose David Jurado Alonso 03 Jul '19

03 Jul '19

Hi, I would like to know if there is any way to obtain a preview image of the video intercom (source endpoint) that makes the call to be shown on the mobile (target endpoint). In this way, the target of the call can see who is calling before answering. intercom (with camera) <--> FS <--> mobile (linphone) Thanks. José David Jurado Alonso

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Installing freeswitch on windows 10
by zoya Ferdowsian 27 Jun '19

27 Jun '19

Hello I am trying to install it on windows 10 I have followed https://freeswitch.org/confluence/display/FREESWITCH/Windows , In the document I see this warning --> On Windows you must Select *CheckOut as-is, Commit as-is* in order to Set core.autocrlf False. >From Git command prompt I run git clone https://stash.freeswitch.org/scm/fs/freeswitch.git C:/FS_GIT/ After completion I type --> *git config --get-all core.autocrlf* returns --> true Where am I supposed to do this to achieve false —> On Windows you must Select *CheckOut as-is, Commit as-is* i Thanks

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BLF stays on
by Sean Devoy 27 Jun '19

27 Jun '19

Hi, This may be a problem specific to our Fanvil phones, but I will throw it out here and hope. I have BLF working except when I call one of the extensions that I have a BLF for (or they call me). Sometimes the BLF stays on after the call. If they use that extension again, the BLF "resynchs" and works fine. I am pretty sure that is the only time the BLF gets "stuck", but I can't say with 100% certainty. FS 1.8 not NATed offsite. Phones are NATed. Thanks, Sean

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Audio fax via FreeSWITCH in Default media mode
by Loi Dang 27 Jun '19

27 Jun '19

Hi all, i'm using FS in the middle of a fax call in Default media mode without loading mod_spandsp.so `t38-passthru` profile param is enabled, and `fax_enable_t38` channel variables are exported. `Asterisk SendFax() -> FreeSWITCH -> Asterisk ReceiveFax()` call flow applied, the codec ulaw was used in all cases All Faxes are succeeded if T.38 involved but when either party is configured intentionally NOT to support T.38, g711 fax was used as expected but nearly all faxes failed (a very few fax calls success in audio mode). I tried removing FS out of the path and the g711 faxes was working fine directly. I'm not sure why so i tweaked some rtp options in FS without knowing very well what they do <param name="auto-rtp-bugs" value="clear"/> <param name="rtp-autofix-timing" value="false"/> <param name="rtp-rewrite-timestamps" value="false"/> when i tried `<param name="rtp-notimer-during-bridge" value="true"/>`, the audio faxes has higher rate of success, but the T.38 failed completely. My version is 1.6.17, tried 1.8.6 without any luck any idea or experience for making g711 fax work with FS in my case? thanks thanks Also tried Proxy media mode to let FS stayed in the path but do not touch the stream,, g711 faxes was working properly, so i guess there was some tweaks by FS default media mode into the rtp stream causing the g711 fax failure rgds, Loi Dang

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FANVIL Phones
by Sean Devoy 26 Jun '19

26 Jun '19

Hi, We have been testing (and using) FANVIL phones. They have been very well received. I am not CONFLUENCE savvy, could someone update the phones page to include this data? Manufacturer Model Version Registration Caller ID Call in/out Call waiting Tranfer Park MWI SRTP TLS Video SIMPLE SLA BLF Fanvil X4/X4G X4-2.10.0.6586 Yes Yes Yes Yes Yes N/A Yes Unknown Unknown Unknown Unknown TCP Only TCP Only Fanvil X5S X5-1.10.13 Yes Yes Yes Yes Yes N/A Yes Unknown Unknown Unknown Unknown TCP Only TCP Only Fanvil X6 X6-V1.3-1.10.13 Yes Yes Yes Yes Yes N/A Yes Unknown Unknown Unknown Unknown TCP Only TCP Only Fanvil X210 X210-1.10.13 Yes Yes Yes Yes Yes N/A Yes Unknown Unknown Unknown Unknown TCP Only TCP Only I hope that survives email transport. Everything seems to work with UDP except BLF and SLA. Thanks, Sean

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freeswitch package on ARM (raspberry pi)
by Bipin Patel 26 Jun '19

26 Jun '19

hi, i was wondering if FS packages will be available for ARM raspbian because we have been running multiple SOHO PBX using rpi for clients for quiet some time now, currently we build from master on raspbian stretch as well as jessie and it works well. We did some stress test with 50 calls and 100 channels using rpi 3b+ with g711u and it works flawless as well as we have it on rpi 2 also easily able to handle 20+ calls and now with the rpi 4 im sure it can handle a lot more easily so why not officially create packages for ARM rather than have to compile from master. We also developed a custom CDR portal we supply with it which gives the user the ability to create extenions, gateways etc etc as xml rather than use a DB or xml curl etc to keep the resource consumption low. Regards, Bipin

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DNS resolving for REGISTER requests
by Sebastian Denz 26 Jun '19

26 Jun '19

Hello, i am using FreeSWITCH as SBC to our SIP provider (Deutsche Telekom in germany). This works quite well, but ~ once in a month we are unable to do outbound calls. It turned out, that the provider changes some IPs from time to time.. That leads to FreeSWITCH being registered to the old IP while INVITE-requests of new dialogs are sent to the new address. The provider terminates this with 403 - forbidden. The registration happens every 5 minutes, but FreeSWITCH does not update the dns cache for the target IP address. So i gets stuck in that situation being registered on the old IP while sending INVITEs to the new one.. The only solution to that is reloading the profile or restarting FreeSWITCH... Is there an option to make FreeSWITCH asking the dns _before every_ REGISTER? Or do you have any other suggestion on how to fix this? Best regards, Sebastian Denz

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mod_fifo and esl application : answer a call with given uuid
by Marco Foschian 26 Jun '19

26 Jun '19

Hello, I'm writing a node.js / esl monitor for a fifo queue that show on a web page the incoming calls and their status (waiting, answered, ringing, etc). The page is used by some operators with a phone registered as member of the queue: they can see who is calling (the number is decoded in a caller-name by the application) before answering the call. Is there a way for the user of the webpage to choose the ringing call to pick-up instead of the oldest one ? When the operator click on the page there is an HTTP call to the node.js server where I have the esl connection and the info of each channel involved in the queue. I already tried some uuid_bridge commands, or "fifo myqueue out nowait" and uuid_setvar fifo_bridge_uuid combinations, but none works. It's possible to implement such a feature with mod_fifo ? There is any alternative ? Best regards, Marco Foschian

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