June 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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new install Debian 9 and FS 1.8 - Seg Fault
by Sean Devoy 24 Jun '19

24 Jun '19

Hi, I have a brand new install of Debian 9 and FS 1.8 (dist) fs_cli -rRS results in error connecting freeswitch -c ends with: Adding API Function 'opus_debug' Segmentation fault (core dumped). This is my first attempt at FS on a nat'ed PC. I ran Debian install I ran FS dist install. I changed Password. NOTHING else. What did I miss? Thanks, sean

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No outbounds calls
by paolo.visnoviz＠vipcomputers.it 21 Jun '19

21 Jun '19

Ok guys, I give up. So I must ask help. :-( In summary: I configured two endpoints and I can call from one to the other. I can receive outbond calls too, through my gateway, and the registration to it seems fine. But I can't call outside. The outbound calls don't works. I don't know where I'm wrong, is there anyone who can put me on the right path, please? This is my gateway conf: <include> <gateway name="outbound-my_provider"> <param name="username" value="123456789"/>   <param name="realm" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="caller-id-in-from" value="true"/> <param name="proxy" value="123.ims.my_provider.net:5060"/> <param name="register" value="false"/>  <param name="context" value="public"/> <param name="extension" value="123456789"/> <param name="extension-in-contact" value="true"/> <param name="expire-seconds" value="3600"/>   </gateway> <gateway name="123456789"> <param name="username" value="123456789(a)ims.my_provider.net"/> <param name="password" value="my_password"/> <param name="extension" value="123456789"/> <param name="proxy" value="ims.my_provider.net"/> <param name="from-user" value="123456789"/> <param name="from-domain" value="ims.my_provider.net"/> <param name="register-proxy" value="123.ims.my_provider.net:5060"/> <param name="expire-seconds" value="1800"/> <param name="retry-seconds" value="120"/> <param name="register" value="true"/> <param name="dtmf-type" value="rfc2833"/> <param name="register-transport" value="udp"/> <param name="context" value="public"/> </gateway> This is my dialplan:  <include> <extension name="outbound_calls"> <condition field="destination_number" expression="(^\d{5,14}$)"> <action application="set" data="effective_caller_id_name=123456789"/> <action application="set" data="effective_caller_id_number=123456789"/>  <action application="bridge" data="sofia/gateway/outbound-my_provider/$1"/> </condition> </extension> </include> I'm natted and the parts comments out are about various tests. The firewall is opened for 5080 versus freeswitch. My freeswitch local ip is 172.16.16.209. This is my pastebin of my external call attempt: https://pastebin.freeswitch.org/view/05befe95 Thank you in advance. Best regards -- Distinti saluti /Paolo Visnoviz/

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Send conference commands through Verto's RPC client without moderator privileges
by Anton Olofsson 20 Jun '19

20 Jun '19

Is it possible to make users able to execute commands such as muting themselves through the RPC client in Verto without the user needing moderator privileges? I'm trying to do this in order for the user's mute status to be broadcast to all other users through the LiveArray (and not just having the user disable the streams locally).

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Calls between extensions with no audio on NAT
by Jose Figueroa 19 Jun '19

19 Jun '19

Hello, I'm having a problem with a NAT scenario, I have some pcs (192.168.10.102 and 192.168.10.103) using verto behind a NAT calling to a LAN FS (192.168.10.143). I added a gateway and the callout is working good enough, but the calls between extensions I am not getting audio, I just get the "No audio stun for a long time!". I set in many ways the following: <param name="ext-rtp-ip" value="192.168.30.143"/> <param name="ext-sip-ip" value="auto"/> I tried auto, auto-nat, $${external_rtp_ip} and $${external_sip_ip} respectively, and that one I just added with no luck. What can be the problem with this? Thanks Jose

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