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Running FreeSwitch as Windows Service
by Adelia C. 19 Oct '11

19 Oct '11

Hello, I am building FreeSwitch through VS 2010 (not VC++ Express) and I generate FreeSwitchConsole.exe as a console application. No FreeSwitch.exe is generated. FreeSwitchConsole runs fine as an application but I need to run the SW as a service. I installed Freeswitch as a local service that calls FreeSwitchConsole.exe, and this is the output of "sc qc FreeSWITCH": C:\FreeSWITCH>sc qc FreeSWITCH [SC] QueryServiceConfig SUCCESS SERVICE_NAME: FreeSWITCH TYPE : 10 WIN32_OWN_PROCESS START_TYPE : 2 AUTO_START ERROR_CONTROL : 1 NORMAL BINARY_PATH_NAME : C:\FreeSWITCH\FreeSwitchConsole.exe LOAD_ORDER_GROUP : TAG : 0 DISPLAY_NAME : FreeSWITCH DEPENDENCIES : SERVICE_START_NAME : LocalSystem Trying to start FreeSwitch as a service results in Error 1053 Timeout - Event Viewer is : Timeout (30000 milliseconds) waiting for the FreeSWITCH service to connect. Or: C:\FreeSWITCH>net start freeswitch The service is not responding to the control function. Is it because I am trying to start a console application as a service? Do I need to have a FreeSwitch.exe and if so, how do I build it? Do I need to create a Windows Service wrapper project? I tried using winserv (supposed to start console apps as service), with same result: C:\FreeSWITCH>..\tools\winserv\winserv start FreeSwitch ERROR 1053: The service did not respond to the start or control request in a timely fashion. What am I missing? Thank you. A.C.

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Ring Group with External Number not transferring after timeout
by Ryan How 19 Oct '11

19 Oct '11

Hi, I have a ring group with an external number I've created using bluebox. Sorry I hope I am posting this in the right forum... What happens is, it will ring for 30 seconds, then stop ringing, then rather than transfer to voicemail, it will start ringing again. If I remove the external number from the ring group then it is ok and works as expected. Is bluebox setting this up wrong? Or is there something I need to change or add? XML is below... On occasion it seems to work, maybe if the external number is busy or something? It is funny because this is exactly the same behaviour I got when setting up the same thing with asterisk. Makes me think there is something I'm not getting when putting in external numbers. I'm using the GIT head of freeswitch (and bluebox) as of yesterday. Thanks, Ryan <extension name="main_number_9" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/> <action application="set" bluebox="settingFail" data="continue_on_fail=true"/> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Ring All"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=3000"/> <action application="export" bluebox="settingContinueOnTimeout" data="originate_continue_on_timeout=true"/> <action application="bridge" data="[ignore_early_media=true,leg_timeout=30]user/2001@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]user/2002@$${location_1}:_:[ignore_early_media=true,leg_timeout=30]sofia/gateway/trunk_1/0447614933"/> <action application="answer"/> <action application="voicemail" data="default voicemail_1 2001"/> <action application="hangup"/> </condition> </extension>

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GSMopen builds
by Nandy Dagondon 19 Oct '11

19 Oct '11

hello, referring to the Wiki, there 4 different builds for voice and SMS support - w/ (or w/o) gsmlib, alsa vs portaudio, etc. i want to test using (A) a mobile phone + USB cable + audio cable and later on, (B) w/ an embedded GSM gateway e.g. Mobigater. will one type of build work in both A and B setups? is there any important item to watch? tks, nandy

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Follow me - 2nd number rings but silence on both ends
by Lars Zeb 18 Oct '11

18 Oct '11

http://pastebin.com/9PR0yCf8 I setup a follow-me scenario in the dialplan above. It does ring through to the second number, but most times there is silence on both ends. If it matters, the second device is a Blackberry. Any ideas on what might correct this? Thanks, Lars

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Kamailio Dispatcher fails in front of FreeSwitch, help needed.
by Henrik Aagaard Sørensen 18 Oct '11

18 Oct '11

When connecting 2 devices directly to FreeSwitch everything works. When connecting to Kamailio (as dispatcher), some calls fails with "Timeout" and some fails with "Too many loops". This is my kamailio.cfg: http://pastebin.com/KEuZmikh When connected directly to FreeSwitch where everything works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/prkjkb7g When connected to Kamailio where nothing works, this is how my clients are registrered in FreeSwitch: http://pastebin.com/b5x6NZyv 215.197.145.132 (sip.my-domain.com) is my Kamailio. pbx.my-domain.com is my FreeSwitch. 96.224.14.164 is my IP at home, where my 2 devices are from. Can anyone help me out? I'm kinda lost at this point. And I don't know if the error is on FreeSwitch or Kamailio (or both).

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Lock down default config
by Ben Naylor 18 Oct '11

18 Oct '11

Hi All I am going live with a Freeswitch server in a couple of weeks, and it will be using a public IP. I am a bit concerned that I may not have tightened security on the box enough to protect it (as best I can) from attack. I have had a look online for something similar to this but a freeswitch version, but can't find anything - http://blogs.digium.com/2009/03/28/sip-security/ Can anyone suggest a list of things I should check before I go live with this server? I have already changed the default password, removed the default user accounts/passwords and removed the external SIP account. Any ideas are greatly appreciated, also any links to websites that I may not have found yet. Thanks for your help! Kind regards Ben

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Loop XML using call_timeout ..
by Antonio 18 Oct '11

18 Oct '11

Hi, Is it possible to loop a bridge until a global call timeout (or other variable) is reached? In the wiki is this example is a fixed number of loops... <extension name="3-way gateway distro"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/${distributor(3gw)}/$1" loop="3"/> </condition> </extension> My idea is to call a group sequential, but once it reaches the last member it starts over again. so when calling a group, once the timeout for each user in the group is reach it hangups the call (the normal behavior), it jumps again to the begin of the group until a global timeout is reached. For example: <extension name="group"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=60"/> <action application="set" data="leg_timeout=10"/> <action application="bridge" data=""${group_call(group1@default+F)}" loop="3"/> </condition> </extension> In my group i have two members only. -- Un cordial saludo / Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

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Mismatched video port number in bridge after answer
by Cindy Leung 18 Oct '11

18 Oct '11

Hello all, I'm have 1.0.head (git-e463321 2011-10-05 19-13-21 -0300) installed and running. I am able to get video both ways if I send the call straight to bridge, like this. <extension name="name1"> <condition field="destination_number" expression="^2015"> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/${use_profile}/2001:2001@dst.proxy.com"/> </condition> </extension> But I'm not getting video in either direction (audio is fine) if I want to get digits from the caller (i.e, a number that is going to be the final destination number.) <extension name="name2"> <condition field="destination_number" expression="^(2014)$"> <action application="answer"/> <action application="sleep" data="1500"/> <action application="spandsp_start_dtmf" /> <action application="read" data="4 5 ivr/Welcome.wav conf_id 10000 #"/> <action application="spandsp_stop_dtmf"/> <action application="set" data="bypass_media=true"/> <action application="bridge" data="sofia/${use_profile}/${conf_id}(a)dst.proxy.com"/> </condition> </extension> Here's the tshark capture. I'm sorry if this is a repeat. I tried to search the mailing list but I couldn't find anything. Any suggestion is welcome. TIA!! 1 <caller's IP> -> <freeswitch IP> SIP/SDP Request: INVITE sip:2014@my.freeswitch.com, with session description c= contains caller's IP m=audio 26646 RTP/AVP 0 8 18 101 m=video 30114 RTP/AVP 123 124 2 <freeswitch IP> -> <caller's IP> SIP Status: 100 Trying 3 <freeswitch IP> -> <caller's IP> SIP/SDP Status: 200 OK, with session description 4 <freeswitch IP> -> <caller's IP> SIP/SDP Status: 200 OK, with session description c= contains FS's IP m=audio 21098 RTP/AVP 0 101 m=video 28620 RTP/AVP 123 5 <caller's IP> -> <freeswitch IP> SIP Request: ACK sip:2014@<freeswitch IP>:5060;transport=tcp 6 <caller's IP> -> <freeswitch IP> SIP Request: ACK sip:2014@<freeswitch IP>:5060;transport=tcp 7 <freeswitch IP> -> <dst proxy ip> SIP/SDP Request: INVITE sip:2001@dst.proxy.com, with session description c= contains FS's IP m=audio 21402 RTP/AVP 0 8 101 13 m=video 22854 RTP/AVP 98 8 <dst proxy ip> -> <freeswitch IP> SIP Status: 100 Giving a try 9 <dst proxy ip> -> <freeswitch IP> SIP Status: 180 Ringing 10 <dst proxy ip> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description c= contains dst proxy's IP m=audio 29462 RTP/AVP 0 8 101 m=video 26506 RTP/AVP 98 11 <freeswitch IP> -> <dst proxy ip> SIP Request: ACK sip:10.10.0.12:5060 12 <freeswitch IP> -> <dst proxy ip> SIP/SDP Request: INVITE sip:10.10.0.12:5060, in-dialog, with session description c= contains the caller's IP m=audio 26646 RTP/AVP 0 8 101 m=video 22854 RTP/AVP 98 13 <dst proxy ip> -> <freeswitch IP> SIP Status: 100 Giving a try 14 <dst proxy ip> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description c= contains dst proxy's IP m=audio 29462 RTP/AVP 0 8 101 m=video 26506 RTP/AVP 98 15 <freeswitch IP> -> <dst proxy ip> SIP Request: ACK sip:<some ip>:5060 16 <freeswitch IP> -> <caller's IP> SIP/SDP Request: INVITE sip:2060@<some ip>:40731;transport=TCP, in-dialog, with session description c= contains dst proxy's IP m=audio 29462 RTP/AVP 0 8 101 m=video 28620 RTP/AVP 123 17 <caller's IP> -> <freeswitch IP> SIP Status: 100 Giving a try 18 <caller's IP> -> <freeswitch IP> SIP/SDP Status: 200 OK, with session description c= contains the caller's IP m=audio 26646 RTP/AVP 0 8 101 m=video 30114 RTP/AVP 123 19 <freeswitch IP> -> <caller's IP> SIP Request: ACK sip:2060@<some ip>:40731;transport=TCP 20 <caller's IP> -> <freeswitch IP> SIP Request: BYE sip:2014@<freeswitch IP>:5060;transport=tcp 21 <freeswitch IP> -> <caller's IP> SIP Status: 200 OK 22 <freeswitch IP> -> <dst proxy ip> SIP Request: BYE sip:<some ip>:5060 23 <dst proxy ip> -> <freeswitch IP> SIP Status: 200 OK Cindy

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X-Headers on BYE- How to save?
by Avi Marcus 18 Oct '11

18 Oct '11

I have a carrier supplying the LRN in the BYE / OK to my BYE. It SEEMS that when the other party hangs up first, I get a sip_h_X-LRN in my xml_cdrs, but not when my FS hangs up first. Presumably this is because their bye happens before the reporting begins, but my bye starts reporting before the OK to the bye is received. So, how would I be able to save access this header? A hangup hook of some sort.. well, that won't have access either. Any ideas? Thanks, -Avi

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External number starts ringing again after rejecting call until timeout
by Ryan How 17 Oct '11

17 Oct '11

Hi, I've got an extension set up to redirect to my mobile. It all works fine, except if I reject the call on my mobile it starts ringing again. It just keeps going until the timeout and then transfers to voicemail. How can I make it go to voicemail if I reject the call on my mobile? I tried adding fail_on_single_reject=true but it didn't seem to do anything. Sorry for the newbie question. I'm slowly learning :). XML is below. Thanks! Ryan <extension name="main_number_11" continue="true"> <condition field="destination_number" expression="^3101$"> <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/> <action application="set" bluebox="settingFail" data="continue_on_fail=true"/> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="set" data="fail_on_single_reject=true"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=Ryan Work Mobile"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=3101"/> <action application="bridge" data="{ignore_early_media=true}sofia/gateway/trunk_1/0447614933"/> <action application="answer"/> <action application="voicemail" data="default voicemail_1 2001"/> <action application="hangup"/> </condition> </extension>

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