October 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Auth and DNS load balancing.
by Sharath Kumar 09 Oct '17

09 Oct '17

Hello, I have the below scenario. FS talking to a service provider and the sip server has DNS load balancing. (FS) INVITE --> abc.com( ip = 10.10.10.10) 407 <------abc.com Ack-----------> abc.com INVITE(credentials)-->abc.com (ip = 11.11.11.11) 407<-------------------abc.com Ack------------> abc.com The session is destroyed at this point and Freeswitch does not resend the INVITE. I want the FS not to give up and keep the INV/407/ACK cycle going. The sofia traces indicate session terminated [904]. I understand that if DNS is going to load balance for every INVITE then the call will never be established. But I still would like to see if it can be done? Anyone else run into this issue ? thank you, Shaks.

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Issue with database updates
by Rick Jarvis 09 Oct '17

09 Oct '17

Is there any reason why FS iterates through all the fields in the channels table, when it updates a field? I’ve created some custom columns (as per https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data… <https://freeswitch.org/confluence/display/FREESWITCH/Variable+presence+data…>) but the values I’m writing to my new columns are being overwritten when FS changes the channel state - it appears from the logs that this is due to FS reading the values from the table, and then setting every column (including my custom fields) when it then updates … to me this is counter-intuitive as far as SQL goes!?

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session id (rfc7329) support (VoIP and Verto)?
by Mundkowsky, Robert 09 Oct '17

09 Oct '17

Does FreeSWITCH have session id (rfc7329) support (VoIP and Verto)? Robert ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________

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PLAYBACK_STOP event not sent trhough ESL for playback if speed has been changed with uuid_fileman
by Harangozó, László 09 Oct '17

09 Oct '17

Hi guys, Setup: Inbound calls are managed in a separate app through an ESL connection, so the FS dialplan is simple: <extension name="CTI_control"> <condition regex="any"> <regex field="destination_number" expression="^(.*)$" /> <action application="socket" data="127.0.0.1:8084 async full" /> </condition> </extension> If a playback operation is started (ESL ExecuteApplication playback) for such a call then both PLAYBACK_START and PLAYBACK_STOP events are sent for the playback application. Except for the case when during play the speed is changed, for example with the following bgapi call: uuid_fileman 8bca0318-a5f5-4170-b82a-d7ab09d018d4 speed:+1 When the speed is changed in such a way, the PLAYBACK_STOP event won't be sent. Environment: Platform: Windows Branch: v1.6 File being played: Issue present with all sound files I have tried The issue happens in a consistent way (all the time when the described scenario happens) Does anyone have a clue, how to get FS to send the PLAYBACK_STOP events?

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Any way to configure failover for shoutcast moh streams?
by Nathan Neulinger 09 Oct '17

09 Oct '17

I know with the plain file MOH you can point to a directory and have it shuffle - is there any way to do that with streams or to have it fail over from a stream to a static file? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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RRNoise - would be pretty cool to see a plugin for this in FS such as for the conference module
by Nathan Neulinger 09 Oct '17

09 Oct '17

https://tech.slashdot.org/story/17/10/01/2014216/donate-you-noise-to-xiphmo… https://people.xiph.org/~jm/demo/rnnoise/ -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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bad ringback tone if I use OPUS
by Alexander Haugg 09 Oct '17

09 Oct '17

Hi All, for outbound calls I have configured following two lines in the dialplan: <action application="set" data="ringback=%(2000,4000,440.0,480.0)" />  <action application="set" data="instant_ringback=true" /> For the G711 and the G722 Codecs the ringback works fine but with the OPUS codec there is a bad noice after the first ringback tone hearable. This problem is veryfied with Bria 4 and the ICE Link Stack Have anyone an idea? Thanks a lot!

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Video call issue from webrtc to sip
by Ram 09 Oct '17

09 Oct '17

Hi, Configured freeswitch as a media gateway between sip and web clients. When video call initiated from Sip client to web-rtc clients, audio sdp is dropped when responding 200 ok to sip client after getting 200 ok response from webrtc clients. where as video sdp is fine in 200 ok. >From the logs found "marking rejected media" might causing issue. but not getting clue for what is causing issue for rejected media. Enclosed complete log for reference. soa_static.c:1189 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): generating local description soa_static.c:1230 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): upgrade with remote description soa_static.c:1264 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): marking rejected media soa_static.c:1029 soa_sdp_mode_set() soa_sdp_mode_set(0x7ff8d9924960, 0x7ff8b0b7f8e0, ""): called soa_static.c:1446 offer_answer_step() soa_static(0x7ff8b0038c40, soa_generate_answer): storing local description soa.c:1730 soa_activate() soa_activate(static::0x7ff8b0038c40, (nil)) called soa.c:1270 soa_get_local_sdp() soa_get_local_sdp(static::0x7ff8b0038c40, [(nil)], [0x7ff8d9926ad0], [0x7ff8d9926acc]) called tport.c:3257 tport_tsend() tport_tsend(0x7ff8b0004930) tpn = UDP/ 5.2.2.6:5060 tport.c:4046 tport_resolve() tport_resolve addrinfo = 5.2.2.6:506 Regards Raman

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Delay Bye Message
by Alexander Perkins 07 Oct '17

07 Oct '17

Hi. I have what appears to be an unusual request. One of my clients has a clause in his contract with the local telco/sip provider that requires the majority of calls to be at or greater than 12 seconds. On a former platform, he has an option called 'Delayed Bye Message.' This option, if checked, allows him to specify how many seconds to delay hanging up the call once leg B initiates the request. So, all calls will last at least 12 seconds. Does anyone know how to do that with Freeswitch? Any help is greatly appreciated. Thank you, Alex Perkins

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Fusion pbx
by Siju Nair 06 Oct '17

06 Oct '17

Hi team I need to install fusion pbx, since am new to this I don’t whether I need to install freeswitch separately or while installing fusion pbx, FS also will be installed... plz help !! Sent from my iPhone

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