October 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

record DID called in DB
by jungle boogie 31 Oct '16

31 Oct '16

Hi All, With the example CDR sql here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/scripts/sq… Inbound destination calls will show the endpoints/call groups that rings, like this (destination_number) field: "2003" "2002" "2600" Is it possible to show the DID that was called as a separate record? example: "2003","12134123000" "2002","12134123000" "2600","12134123000" In this case, 12134123000 is a DID setup on the freeswitch server that's called and it rings extensions/call groups 2003, 2002 and 2600. I see destination_number is used on outbound calls to record what extension and/or DID was dialed. Thanks, sean

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testing again
by Ken Rice 30 Oct '16

30 Oct '16

Is it working?

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Re: [Freeswitch-users] stash down?
by David Villasmil 30 Oct '16

30 Oct '16

Nevermind, it's back up. ᐧ On Sun, Oct 30, 2016 at 9:03 AM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > Hello guys, > > I'm trying to push to stash but i keep on getting > > Service Temporarily Unavailable > > The server is temporarily unable to service your request due to > maintenance downtime or capacity problems. Please try again later. > ------------------------------ > Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 > > ᐧ >

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stash down?
by David Villasmil 30 Oct '16

30 Oct '16

Hello guys, I'm trying to push to stash but i keep on getting Service Temporarily Unavailable The server is temporarily unable to service your request due to maintenance downtime or capacity problems. Please try again later. ------------------------------ Apache/2.2.22 (Debian) Server at freeswitch.org Port 443 ᐧ

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What concepts in FreeSWITCH did you struggle with when you first started?
by Brian West 27 Oct '16

27 Oct '16

I would like to get community input on what concepts caused you the most headache and any input/ideas that would help me design some tutorials / videos that would help new users coming to FreeSWITCH. Your feedback will be greatly appreciated. Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Can't build 1.6 on Jessie w/ SCTP
by Josh Jore 27 Oct '16

27 Oct '16

Hi, I just downloaded a fresh Jessie64 and failed building Freeswitch v1.6 with SCTP. Seems maybe SCTP just isn't supported right now or ...? Are there any configurations where this works? $ vagrant init debian/jessie64 $ vagrant up $ vagrant ssh Follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie ... $ git checkout v1.6 # https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/96eca8abb… $ ./bootstrap.sh -j $ ./configure --enable-sctp --prefix=... ... Making all in tport make[8]: Entering directory '/home/vagrant/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' make all-am make[9]: Entering directory '/home/vagrant/freeswitch/libs/sofia-sip/libsofia-sip-ua/tport' LTCOMPILE tport.lo LTCOMPILE tport_logging.lo LTCOMPILE tport_stub_sigcomp.lo LTCOMPILE tport_type_udp.lo LTCOMPILE tport_type_tcp.lo LTCOMPILE tport_type_sctp.lo tport_type_sctp.c: In function 'tport_sctp_init_socket': tport_type_sctp.c:206:10: error: variable 'initmsg' has initializer but incomplete type struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:206:10: error: excess elements in struct initializer [-Werror] tport_type_sctp.c:206:10: error: (near initialization for 'initmsg') [-Werror] tport_type_sctp.c:206:23: error: storage size of 'initmsg' isn't known struct sctp_initmsg initmsg = { 0 }; ^ tport_type_sctp.c:211:36: error: 'SCTP_INITMSG' undeclared (first use in this function) if (setsockopt(socket, SOL_SCTP, SCTP_INITMSG, &initmsg, sizeof initmsg) < 0) ^ tport_type_sctp.c:211:36: note: each undeclared identifier is reported only once for each function it appears in tport_type_sctp.c:206:23: error: unused variable 'initmsg' [-Werror=unused-variable] struct sctp_initmsg initmsg = { 0 }; ^ cc1: all warnings being treated as errors

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Running a script on a completed transfer
by Walter Smith 27 Oct '16

27 Oct '16

I need to record calls if/when they touch a certain set of extensions. I have most of the logic working but am baffled by one thing: how do I cause an application (it’s a lua script) to run when a channel is bridged as the result of a transfer? My usual bag of tricks isn't working because the bridge happens "magically" in sofia in response to the REFER. For example, execute_on_answer doesn’t help because the answer happens during the attended transfer -- I need to run AFTER the transfer (that is, when the original A leg is bridged to the attended transfer’s B leg). Example scenario: 1. External call comes in (channel A = PSTN to switch, channel B = switch to phone with extension 1000) 2. Ext 1000 does an attended transfer to ext 1001 (channel C = 1000 to switch, channel D = switch to 1001) 3. Transfer completes (sofia bridges channel A to channel D) I want my script to run on either channel A or D after 3 happens. What’s the best way to do this? Thanks, Walter

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ClueCon Weekly - Ken's mic volume
by Valter Nogueira 27 Oct '16

27 Oct '16

I feel that Ken's mic is a little low comparing to other's ones. It could not be a problem for native speakers, but it is a little hard for me to get what he is talking. Regards, Valter

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dtmf mode = none
by bugi＠nicelan.tk 26 Oct '16

26 Oct '16

Hello FS Experts, I need to disable dtmf tones on an active call in FS. My case scenario is: regular call is setup by caller -> FS -> outgoing trunk and dtmf codes should be working fine. After some time a dtmf blocking event will be send by lua script or FS command line. This should block sending dtmf codes for caller and makes them "invisible" on outgoing trunk. Is this possible to do this with FS? I was thinking that maybe I can change dtmf_mode during active call to none, but this doesn't seem to be working. Any ideas? Best wishes, bugi

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SRTP support
by Tom Chen 26 Oct '16

26 Oct '16

Hello, By default, Freeswitch seems always translates incoming SRTP packets to outbound RTP only packets as the INVITE SIP message to Receiver from FS contains only RTP/AVP audio media in SDP section even if the Caller's INVITE has RTP/SAVP audio media only. Caller to FS Session Initiation Protocol (INVITE) . . . . Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 1001 625 1382 IN IP4 10.113.49.31 Session Attribute (a): rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics Media Description, name and address (m): audio 7078 RTP/SAVP 96 97 98 99 0 8 101 100 102 FS to Receiver Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): FreeSWITCH 1477472443 1477472444 IN IP4 172.17.10.23 Media Description, name and address (m): audio 25206 RTP/AVP 0 8 101 13 Is there a way to let FS send SRTP packets to receiver as the receiver can accept SRTP media only? Tom

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