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Re: [Freeswitch-users] how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz)
by ax lyb 16 Jan '15

16 Jan '15

hi, brian,Andrew i known a little in lua,i do my customer callcenter in the following style: 1. create a inbound connect to see what channel hangup (BaseFreeswitchClient is a socket-connection wrap) int connectFS() { //"auth ClueCon\n\n"; const char* AUTH_ME="ClueCon"; m_pFsClient = new BaseFreeswitchClient(reinterpret_cast<IChannelDataCallback*>(this)); if(m_pFsClient->ConnectFS(AUTH_ME) < 0 ) { delete m_pFsClient; m_pFsClient = NULL; return -1; } // const char* strFilter = "event plain CHANNEL_CREATE \ // CHANNEL_ANSWER \ // CHANNEL_BRIDGE \ // CHANNEL_HANGUP \ // CHANNEL_HANGUP_COMPLETE\n\n"; const char* strFilter = "event plain CHANNEL_HANGUP_COMPLETE\n\n"; if( m_pFsClient->filterEvent(strFilter) < 0 ) { delete m_pFsClient ; m_pFsClient = NULL; return -1; } return 0; } 2. create a server for outbound connect, the outbound connect for every call-session to FILTER CHANNEL_ANSWER,CHANNEL_HANGUP_COMPLETE,CHANNEL_EXECUTE_COMPLETE case ESL_EVENT_CHANNEL_DATA: { const char* strEvtName = evt.getHeader("Event-Name"); char szFilter[1024]; const char* strFilter = "event plain CHANNEL_ANSWER CHANNEL_BRIDGE CHANNEL_EXECUTE_COMPLETE DTMF\n\n"; snprintf(szFilter, sizeof(szFilter),"%sfilter Unique-ID %s\n\n",strFilter, evt.getHeader("Caller-Unique-ID")); int nWriteSize = strnlen(szFilter,sizeof(szFilter)); LOG(LS_ERROR)<<strEvtName << "," <<evt.getHeader( "Caller-Unique-ID") <<",nWriteSize="<<nWriteSize <<","<<szFilter; p->onWrite(szFilter, nWriteSize); } 3. set "session.SetAutoHangup(false);”, which channel_variable should i set? thanks > 在 2015年1月16日，08:18，freeswitch-users-request@lists.freeswitch.org 写道： > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: Help Needed Debugging Lua Script (Luis Daniel Lucio Quiroz) > 2. Re: how can i keep A-LEG do not hangup (Luis Daniel Lucio Quiroz) > > 发件人: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> > 收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 日期: 2015年1月16日 GMT+808:13:15 > 回复－收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 主题: 回复： [Freeswitch-users] Help Needed Debugging Lua Script > > > Don't loose your time, contact me offline > > On Jan 15, 2015 5:49 PM, "Sina Owolabi" <notify.sina(a)gmail.com <mailto:notify.sina@gmail.com>> wrote: > Hi List, > > I think I have finally come up with something that works, the script captures the details correctly, but I wonder if there is a better way to write it? I am trying to get it to call the dialer back and play a message, which it does correctly. > Please what can I do to make it to retry the dialler's number in case the call doesn't go through the first (say) two times, just in case the dialler cut it off mistakenly, or the telco plays a repeated message after freeswitch hung up, preventing the callback from coming through? (I've seen this happen a few times with a local telco). Thanks! > > number_to_call = argv[1]; > caller_id = argv[2]; > api = freeswitch.API(); > dialString = "{origination_caller_id_name="..caller_id..",origination_caller_id_number="..caller_id.."}sofia/gateway/sipgw/"..number_to_call..""; > session1 = freeswitch.Session(dialString); > session1:sleep(5000); > > if (session1:ready()) then > session1:sleep(35000); > api:execute("bgapi originate", "session1"); > session1:sleep(3000); > session1:streamFile('/tmp/stop_calling_me_stalker.wav'); > session1:hangup("NORMAL_CLEARING"); > end > > > On Wed Jan 07 2015 at 2:45:04 PM Avi Marcus <avi(a)avimarcus.net <mailto:avi@avimarcus.net>> wrote: > Two things: > 1) You aren't grabbing the arg, but the channel variable.. try this in your script: > caller_id_number = argv[1] > number_to_call = argv[2] > > 2) I don't think you're managing your hangup/callback originate properly. > I don't think you want to use bgapi... or maybe you just need a destination. It's "originate <http://wiki.freeswitch.org/wiki/Mod_commands#originate> sofia/A endpoint" -- you need to specify where it goes to, the lua script can't "receive" the call. You can have it received by e.g: &lua(pickup.lua) > > api = freeswitch.API() > api:execute("originate", DialString.." &lua(pickup.lua)"); > > Also: > Maybe you want to use it as a hangup hook. Instead of: > <action application="lua" data="callback.lua ${effective_caller_id_number} ${destination_number}"/> > Do: > <action application="set" data="api_hangup_hook=lua callback.lua ${effective_caller_id_number} ${destination_number}"/> > <action application="hangup" data="486"/> > > -Avi > > On Wed, Jan 7, 2015 at 1:52 AM, Sina Owolabi <notify.sina(a)gmail.com <mailto:notify.sina@gmail.com>> wrote: > Hi List, > > FreeSWITCH newbie here again. > I am trying to cobble togther a lua callback script, my first attempt > was successful, but I am trying to make it slightly more elegant. > I don't see any errors when I try to run this but the callback isnt happening. > This is my very second attempt trying to write in lua, so I would be > very grateful for any help. > > The user is expected to dial in, have the call hangup and FreeSWITCH call back. > > I'm passing a modified $effective_caller_id_number and > $destination_number to the lua script: > > <extension name="callyouback"> > <condition field="${caller_id_number}" > expression="^1(\d{10})$"require-nested="false"> > <action application="set" data="effective_caller_id_number=+234${1}"/> > <action application="set" data="effective_caller_id_name=+234${1}"/> > </condition> > <condition field="destination_number" expression="^012345(6)(7)$"> > <action application="set" data="destination_number=+12312345${1}${2}" /> > <action application="lua" data="callback.lua > ${effective_caller_id_number} ${destination_number}"/> > <action application="gentones" data="%(3000,0,430,450,550)"/> > <action application="hangup" data="NORMAL_CLEARING"/> > </condition> > > > The script itself: > > api = freeswitch.API(); > call_string = "bagpi originate > {origination_caller_id_name="..caller_id_name..",origination_caller_id_number="..caller_id_number.."}sofia/gateway/mysipgate/"..number_to_call.."" > > freeswitch.msleep(5000); > if (session:ready()) then > caller_id_number = session:getVariable("destination_number"); > caller_id_name = session:getVariable("destination_number"); > number_to_call = session:getVariable("effective_caller_id_number"); > > api:executeString(call_string); > freeswitch.msleep(2000); > session:streamFile("/tmp/get_off_my_lawn.wav"); > session:hangup("NORMAL_CLEARING"); > end > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > 发件人: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> > 收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 日期: 2015年1月16日 GMT+808:18:10 > 回复－收件人: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > 主题: 回复： [Freeswitch-users] how can i keep A-LEG do not hangup > > > Read about queues > > On Jan 15, 2015 11:53 AM, "Brian West" <brian(a)freeswitch.org <mailto:brian@freeswitch.org>> wrote: > Thats not relevant in this case, The autoHangup is when used in say lua... > > This case you will probably want this combo: > > https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_e… <https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket#mod_e…> > https://wiki.freeswitch.org/wiki/Variable_park_after_bridge <https://wiki.freeswitch.org/wiki/Variable_park_after_bridge> > > > > > On Thu, Jan 15, 2015 at 7:27 AM, Andrew <aademattia(a)comcast.net <mailto:aademattia@comcast.net>> wrote: > This may be able to help you but I set session.SetAutoHangup(false); > > This will stop the dial plan from ending. > > <> > From: freeswitch-users-bounces(a)lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of ax lyb > Sent: Thursday, January 15, 2015 2:52 AM > To: freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: [Freeswitch-users] how can i keep A-LEG do not hangup > > > > All: > > recently i write a small program based freeswitch, in this program i only do things like callcenter mod, > > here is what i do: > > > > 1. originate a call (“ originate sofia/gateway/gw01/xxxx my-out-call XML default”), in the default.xml, > > i already config it as follow: > > > > <extension name=“my-call-out"> > > 65 <condition field="destination_number" expression="^my-call-out$"> > > 66 <action application="socket" data="127.0.0.1:9600 <http://127.0.0.1:9600/> async full"/> > > 67  > > 72 </condition> > > 73 </extension> > > > > 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , > > bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, > > these steps run normally ok. > > > > 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, > > i’ll leave the other call do nothing,for i can not get more free agents to service; some special scene > > is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only > > A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it’s not user hangup manually), > > how can i do make the C3 call not hangup until i get a free agent to service it? > > > > > > any suggest is appreciate, > > > > > > > > ax.lyb.lei(a)gmail.com <mailto:ax.lyb.lei@gmail.com> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > -- > Brian West > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com <http://www.freeswitchbook.com/> > http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> > T:+19184209001 <tel:%2B19184209001> | F:+19184209002 <tel:%2B19184209002> | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org ax lyb ax.lyb.lei(a)gmail.com

1 0

how can i keep A-LEG do not hangup
by ax lyb 16 Jan '15

16 Jan '15

All: recently i write a small program based freeswitch, in this program i only do things like callcenter mod, here is what i do: 1. originate a call (“ originate sofia/gateway/gw01/xxxx my-out-call XML default”), in the default.xml, i already config it as follow: <extension name=“my-call-out"> 65 <condition field="destination_number" expression="^my-call-out$"> 66 <action application="socket" data="127.0.0.1:9600 async full"/> 67  72 </condition> 73 </extension> 2. when call-out-destination customer pick up the call, i control the flow to playback a wav file, then , bridge the call to a free agent while the wav file play over in CHANNEL_EXECUTE_COMPLETE, these steps run normally ok. 3. the question is : if i have two agent in one moment, and this time i have 3 or > 3 out-call to bridge, i’ll leave the other call do nothing,for i can not get more free agents to service; some special scene is i have two agents (A1,A2) service 3 call (C1,C2,C3), A1 service C1, A2 service C2,then C3 is only A-LEG when play over a wav file,about 100 seconds later, C3 is hangup(it’s not user hangup manually), how can i do make the C3 call not hangup until i get a free agent to service it? any suggest is appreciate, ax.lyb.lei(a)gmail.com <mailto:ax.lyb.lei@gmail.com>

4 3

Make conference play command async based on when prompt finishes
by Ignacio Durli 15 Jan '15

15 Jan '15

Hello, Our application interacts with freeswitch both using api commands and event socket library for receiving events in an IVR application. One of our requirements is to continue with IVR flow once a prompt has finished. Prompts are being played using the "conference <confname> play <file_path> [<async>|<member_id>]". We found that FS response comes immediately after the play starts. Although we found that an event arrives when the prompt ends, we don't find convenient to have that resolved on the code that is receiving the events and would like to have something that returns once the prompt is finished. Is there any way to susbscribe an api call in a way that returns once a specific event happens? What's the difference between using or not the "async" param in the "conference play" api call?

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Sound delays when using WebRTC
by Robert Smallwood 15 Jan '15

15 Jan '15

Hi All, I'm currently using a browser (chrome) to make calls via webrtc to freeswitch (1.5.15b+git~20141219T193610Z~35cb0ad286~64bit). However, I am experiencing delays when hearing our initial prompting message. I'd like to hook onto some event that lets me know that all of session.streamFile() will be heard and not cut off. We have a dial plan, and in that dial plan (js file) we do this sequence of events... (psuedo code) if(session.ready()) session.answer() ...some reading of headers... session.streamFile("Our welcome prompt", func return true) On the client side I can watch the logs and notice that I hear media as soon as this log comes through: | jssip.rtcsession.rtcmediahandler | ICE connection state changed to "connected" However, this can be a second or two behind the streamFile call in the dial-plan. What I would like to be able to do is hook into the ICE state change in the dial-plan file and wait until I get it to start streaming the file. I have tried playing around with some of the methods found here: https://wiki.freeswitch.org/wiki/Session, namely session.mediaReady, session.waitForMedia, and checking the session.state to see if any of them change when I get the ice state message client side to no avail. Is there a way to accomplish this (Listening for ICE state changes in a dial plan)?

2 1

leg_timeout and group_confirm_cancel_timeout
by Ali Pey 15 Jan '15

15 Jan '15

Hello, If I am using group confirm and have a number of destinations to ring and have group_confirm_cancel_timeout set, what happens if my first destination answers the phone but does not press any key to confirm the call. Will the call ever go to the second destination? In another word, when using group confirm, a call passes through three phases: 1- Call is ringing 2- Call is answered, waiting to be confirmed 3- Call is confirmed and bridged Is there any timeout setting for Phase two? Would bridge continue to the next destination if phase 2 fails? Thanks and regards, Ali Pey

1 0

Playing with ACL and authenticate users
by Elhodred 15 Jan '15

15 Jan '15

Hi all, I was playing around trying to have some users authenticated by IP and some users to be challenged for username/password auth. The problem is that I don't get it all. If I configure 2 users in this way: <user id="6001" cidr="192.168.28.12/32"> <params> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I added in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> User 6001 can authenticate if it's using IP 192.168.28.12 and can place calls, but user 6002 is rejected by acl "domains" Then I tried this: <user id="6001"> <params> <param name="auth-acl" value="192.168.28.12/32"/> </params> <variables> <variable name="accountcode" value="6001"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> <user id="6002"> <params> <param name="password" value="111111"/> </params> <variables> <variable name="accountcode" value="6002"/> <variable name="user_context" value="GOLDEN"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> </variables> </user> And I deleted in sofia.conf.xml <param name="apply-inbound-acl" value="domains"/> and added <param name="auth-calls" value="true"/> Now user 6002 can place calls but for user 6001 I see in the log : 2013-05-23 18:15:27.222090 [WARNING] switch_core_state_machine.c:514 bed463c1-742e-4e24-829f-a6188a6667b3 sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX Abandoned 2013-05-23 18:15:27.222090 [NOTICE] switch_core_state_machine.c:517 Hangup sofia/GOLDEN/XXXXXXXX(a)XXX.XXX.XXX.XXX [CS_NEW] [WRONG_CALL_STATE] This is my dial plan config: <context name="GOLDEN"> <extension name="unloop"> <condition field="$${unroll_loops}" expression="^true$"/> <condition field="$${sip_looped_call}" expression="^true$"> <action application="deflect" data="$${destination_number}"/> </condition> </extension> <extension name="outside_call" continue="true"> <condition> <action application="set" data="outside_call=true"/> </condition> </extension> <extension name="hangup"> <condition field="destination_number" expression="^(hangup)$"> <action application="hangup"/> </condition> </extension>  <extension name="remove_prefix_2" continue="true"> <condition field="network_addr" expression="^XXX\.XXX\.XXX\.XXX$"/> <condition field="destination_number" expression="^12345(\d+)$"> <action application="log" data="Removing leading digits"/> <action application="set" data="destination_number=$1"/> </condition> </extension> <extension name="distributor_TEST2_37360______"> <condition field="${sip_authorized}" expression="^true$" break="never"/> <condition field="destination_number" expression="^(\d+)$"> <action application="log" data="Dialing User"/> <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,NO_ROUTE_DESTINATION,UNALLOCATED_NUMBER,407"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="user/$1(a)XXX.XXX.XXX.XXX"/> </condition> </extension> And now I'm lost. Please, can anybody point me where the issue can be? Do I need to have different sofia profiles, one for IP authentication and one for username/password auth? Regards, Alfonso.

6 7

Re: [Freeswitch-users] Security Issue
by Ahmed Habiba 15 Jan '15

15 Jan '15

Thank you really Michael,David and Brian, I did a simple change to the external sip profile which resolved the issue from my point of view. what I did is I add the below line to the external sip profile, which inform it to valid any request from external system against ACL list. <param name="apply-inbound-acl" value="domains"/> From: Michael Collins <msc(a)freeswitch.org <mailto:msc@freeswitch.org>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Date: January 15, 2015 at 7:19:07 AM GMT+3 Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Security Issue On Wed, Jan 14, 2015 at 9:40 AM, Ahmed Habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: Thank you really David, Here is my point, the sip-trace in the first mail shows that, the call comes to public context mainly through port 5080, and however the originator IP was not defined in my ACL list Freeswitch continue to process the call for some reason. Just an FYI, the external profile does not have auth-calls param set to true, so FS simply tries to route the call in the public context without sending back an auth challenge. Since the public context is pretty paranoid it's not exactly easy to dial out. Also, just because FS tries to route the call does not mean that FS considers the call to be "authenticated." If you want all traffic coming in to your server to be authenticated then either send it all to the internal profile (i.e. port 5060) or add auth-calls to your external profile. The bigger question you may want to ask is: why are these random IP even getting to your server? Do you allow public access to your system? If so, why? If not, then you need a firewall (iptables or whatnot) to block those SIP messages from ever getting to your FreeSWITCH. You may also be interested in something like fail2ban and voipbl.org <http://voipbl.org/>. -MC even if it come to 5060, I was expecting some request for digest authentication, which is not shown in the log. From: David Villasmil Govea <david.villasmil(a)gmail.com <mailto:david.villasmil@gmail.com>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Date: January 14, 2015 at 8:30:35 PM GMT+3 Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] Security Issue Authorization is done if you configure your sip profile to do it. By default 5060 (internal) requires authentication, 5080 (external) doesn't but it does use the ACL to allow or not calls.

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Issue with Freeswitch Behind nat
by Frederick Pruneau 15 Jan '15

15 Jan '15

Hi guys, We have an issue with one freeswitch server behind nat. We have a setup like this: -One master Freeswitch server -One freeswitch server connected to the master (Public IP) - Server A -One freeswitch server connected to the master (behind nat) - Server B If server A call server B, nothing happens. There is no sound. After 30 sec, it times out. We have done a tcpdump. From server A to master packets are ok. From Master to server B, we have seen that there is no source and no destination ports for sip invite. If we use our cellphone and we call server B, there is no problem. I have attached the failed call pcap file and freeswitch's log file so you can take a look at them. Master = Freeswitch v1.4.13 Server A = Freeswitch v.1.4.13 Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server) Thanks in advance. PS: The failed call is from 514-448-0773.

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postgresql
by Nick Zaitsev 15 Jan '15

15 Jan '15

Good day to you, Please,advise me, how can i use postgres schema in freeswitch config (cdr_pg_csv.conf.xml)? I'd like to use cdr schema instead public, for example here <param name="db-info" value="host=localhost dbname=freeswitch connect_timeout=10" /> -- Nick Zaitsev

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Miscellaneous question
by John Nash 15 Jan '15

15 Jan '15

I have some points where I request you to please comment... 1- When using sched_hangup to auto hangup call in XX seconds, does freeswitch uses some considerable additional resource? Like may be creating one extra thread to time call or such call limit is anyway part of the design whether someone uses or not. 2- I have read in some articles where it is mentioned that freeswitch is able to bridge media (all RTP packets pass through freeswitch) without degrading "quality". Is it different than using RTPPROXY/mediaproxy better or worse? (Assuming IVR/conference etc are not being used)

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