- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

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[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

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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

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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

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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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Virtual DB FreeSWITCH
by Usama Zaidi 30 Jul '15

30 Jul '15

Hi all, I wanted to know if there's something similar to http://www.opensips.org/About/News0035 available in freeSWITCH, I want to write CDRs etc to DB but my environment doesn't support multicast/floating IPs. Regards. -- I'd love to change the world, but they wont gimme the source code to it

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ignore_early_media=true behaviour
by Vishal Sharma 30 Jul '15

30 Jul '15

Hi All, I see a strange behaviour for ignore_early_media variable. I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number. If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB. <extension name="SIP-TO-SIP"> <condition field="destination_number" expression="^91(\d+)$"> <action application="answer"/> <action application="set" data="hold_music=/srv/sounds/123.mp3"/> <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/> <action application="hangup"/> </condition> </extension> caller ------------------FS-------------------Callee INVITE-------------->| <-------------------100 <-------------------200 ACK-------------------> ###############| INVITE------------------> ###############| <------------------------100 BYE------------------>| <-------------------200| ###############| CANCEL------------------> ###############| <------------------------200 ###############| <---------------------------487 ###############| ACK-----------------------> But when I include ignore_early_media=true in bridge command, Freeswitch ignores BYE sent by legA ,when it is trying to bridge the call with leg B. caller ------------------FS-------------------Callee INVITE-------------->| <-------------------100 <-------------------200 ACK-------------------> ###############| INVITE------------------> ###############| <------------------------100 BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| ###############| <---------------------------486 ###############| ACK-----------------------> <------200 ok BYE I would like to end the call if Leg A disconnects the call while I want to keep ignore_early_media = True so that I can play a hold music file instead of letting legA user listen to call failure message when FS tries multiple numbers Any suggestions .... Thanks a lot Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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recording ring tong
by Pete Kay 30 Jul '15

30 Jul '15

Hi I am trying to record the incoming ring tone. Does anyone know how to do that? The record app does not seem to be able to record ring tone. Thanks, Pete

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FS and T.38 fax
by Mimiko 30 Jul '15

30 Jul '15

Hello. I'm on FS 1.3.13b_cad607d72e. I have two D-Link VoIP gateways DVG-5008S and DVG-2024. Also there are two Canon MFP with faxes. Each DVG have on one port connected a Canon MFP. The FXS ports are configured to use T.38 for fax. Sending fax from one MFP to another starts, but ends with error. Switching ports to use T.30 allows to send and receive fax between MFPs. Doesn't this version of FS support T.38? Does latest version support? Or I miss something in configuration? -- Mimiko desu.

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Bridge incoming call to internal extension and to external gateway
by Vivat 29 Jul '15

29 Jul '15

Hi, I try to ring an incoming call simultanously at my office phone (Snom IP phone) and at my mobile (via a second SIP gateway). Actually it should be simple: bridge with two targets. So I set up in the dialplan something like <action application="bridge" data="user/1000,sofia/gateway/MyTelco/[my mobile no]"/> But the office phone rings only for less then a second and ends then with "Call completed elsewhere". That means I can not answer the call at the office phone. The forwarding to the mobile works perfect. Each single target works (i.e. if I remove the mobile gateway from "bridge" then my office phone rings for longer and I can answer the call there). I read the wiki for the FollowMe dialplan but that is actually not what I want. There should be no first phone with a timeout and then ringing the second phone but both phones should ring in parallel and the phone with the first answer wins. Any hints how to solve this? Thanks in advance.

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