July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Complex phrase macro
by Alex Lake 04 Aug '15

04 Aug '15

I'd like a macro that can play multiple sound files from a specific directory. eg. here's one that should (though I've not tested it!) play 3 sound files: phrase "xplay3:/home/pabx/004-3774/x,001,002,003" <macro name="xplay3"> <input pattern="(.*),(.*),(.*),(.*)"> <match> <action function="play-file" data="$1/$2.wav"/> <action function="play-file" data="$1/$3.wav"/> <action function="play-file" data="$1/$4.wav"/> </match> </input> </macro> What I'd like to do is one that can play an arbitrary number of files.

5 8

webrtc/Sip issue
by Chris Mandra 03 Aug '15

03 Aug '15

Hi guys - one of my users keeps getting this error in fs_cli when he tries to connect via Chrome (web-rtc) on a macbook running yosemite. 2015-07-28 00:42:20.669463 [ERR] switch_core_media.c:6538 AUDIO RTP REPORTS ERROR: [Remote Address Error!] I'm using essentially the same setup and never have an issue. Any ideas? He doesn't get anything very useful in the chrome console, just this: ********************* Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | closing INVITE session edsmilh98mlqt0a48qo0tvmlnm0u3m sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.invitecontext.mediahandler | closing PeerConnection sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.dialog | dialog edsmilh98mlqt0a48qo0qf3p9f1n97g6QD4ZtFHXgrH deleted sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event failed sip-0.6.4.min.js:36 Tue Jul 28 2015 15:22:55 GMT-0400 (EDT) | sip.inviteclientcontext | emitting event rejected sc-webrtc.js:931 _rejected called ******************* Any ideas? We're using latest FS and sip.js thanks, chris

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FreeSWITCH Automated Testing
by Brian West 03 Aug '15

03 Aug '15

FreeSWITCHers, So after a bit discussion, a case of red bull, an approach to automating functional testing has now solidified. Audio Path is verified via DTMF in-band, Video Path is verified by using QR codes. Here are two VERY crude examples: https://www.dropbox.com/s/4fkdcgzo8dt0hdr/conference_test.pl?dl=0 https://www.dropbox.com/s/497rm2hgopp5u7h/eavesdrop_test.pl?dl=0 Input? -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com 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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mod_local_stream moh confusion
by Lappe, Adam 03 Aug '15

03 Aug '15

Hello, i am testing the latest 1.4.19 Version of FreeSWITCH. Currently we are running an old 1.2.7 Version. Everything seems to work fine, but there is 1 error that is very confusing: When a call gets transfered by the callee (i.e. by the receptionist) the call will be terminated. All I see is this error line: [ERR] switch_core_file.c:149 Invalid file format [local_stream] for [moh]! I don't use the local_stream module. freeswitch@internal> module_exists mod_local_stream false This error does not exists with the old version. Is this a bug, or am I missing something? Thanks in advance, Adam

3 10

Unpark call
by Carlos Ruiz Díaz 01 Aug '15

01 Aug '15

Hi, I'm looking for a way to unpark a call remotely. I have this scenario that involves Kamailio as registrar and proxy, and I'm sending calls to FS whenever an user is busy and can't pick up a call. This call is parked by FS. When Kamailio detects the user in question got available, I want to unpark the previous call and connect it to the original destination. Any ideas on how to do this in a clean way? Regards, -- Carlos http://caruizdiaz.com

2 6

group_confirm_file on Specific Legs
by Lyle Pratt 31 Jul '15

31 Jul '15

Hi Guys, We use group_confirm_file with group_confirm_key=exec to require that users "Press ANY key to accept the call" when answering. The problem is, that we only want to do this for calls to external devices. For internal (registered SIP) devices, we do not want to require a keypress to connect. How can we selectively use group_confirm_file on specific bridged legs, while still simultaneously dialing out to all legs? Thanks, Lyle

1 0

[CS_NEW] [WRONG_CALL_STATE]
by Rajil Saraswat 31 Jul '15

31 Jul '15

Hello, I have two freeswitch servers connected over vpn like the following: FS A <> openvpn<> FS B Server A has the ip address 172.16.1.2 Server B has the ip address 192.16.1.2 openvpn gateway is on 10.8.0.1 Both the servers have the common acl defined to not nat the above addresses like following: <list name="nat.auto" default="deny"> <node type="deny" cidr="10.8.0.0/24"/> <node type="deny" cidr="172.16.1.0/24"/> <node type="deny" cidr="192.168.1.0/24"/> </list> <list name="localnet.auto" default="deny"> <node type="allow" cidr="172.16.1.0/24"/> <node type="allow" cidr="10.8.0.0/24"/> <node type="allow" cidr="192.168.1.0/24"/> </list> I am able to call the extensions on server B from server A. However if i try calling the extension on server A from server B i get an error of [CS_NEW] [WRONG_CALL_STATE]. The only way i can make a successful call from B to A is when i add the network of B in the acl on server A as follows: <list name="domains" default="deny"> <node type="allow" domain="$${domain}"/> <node type="allow" cidr="192.168.1.0/24"/> Is there a reason why i need to do this? Thanks Rajil

3 3

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 31 Jul '15

31 Jul '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

Endless playback in conference
by Danny Gershman 30 Jul '15

30 Jul '15

I'm trying to do an endless playback of an mp3 file in a conference. I have a couple of ideas, but none seem really solid. 1) Pass a variable on play and monitor from mod_event_socket and play again if not forcibly terminated. 2) Load up local_stream dynamically from an xmlhttp server, and then restart the local stream service, however will interrupt MOH for other users. Any other ideas? Is there a way to do looping for "api" through mod_event_socket? I know you can with "sendmsg" --Danny Gershman

5 9

ClueCon Extra ticket
by Podrigal, Aron 30 Jul '15

30 Jul '15

Hi All, One of our team is unable to attend so we have an extra ticket to give away. First come first serve. Please contact me in private aron at mongotel.com. -- Aron Podrigal - '1000001', '1110010', '1101111', '1101110' '1010000', '1101111', '1100100', '1110010', '1101001', '1100111', '1100001', '1101100' P: '2b', '31', '33', '34', '37', '34', '35', '38', '36', '30', '39', '39'

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