- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch SIP and firewalls
by Julien Claassen 20 May '08

20 May '08

Hi everyone! Sorry, if this is stupid, but I better ask anyway, because I need someone to configure the firewall for me, as it requires interface-techniques I can't use. Which ports do I have to open up to realise a good SIP connection to the world. I work on a linux-machine behind a netgear router/firewall. the only thing I like to have is SIP. My account is at ekiga.org. If more infomration is required, I'll be happy to supply it. Kindest regards Julien -------- Music was my first love and it will be my last (John Miles) ======== FIND MY WEB-PROJECT AT: ======== http://ltsb.sourceforge.net the Linux TextBased Studio guide ======= AND MY PERSONAL PAGES AT: ======= http://www.juliencoder.de

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sip profiles
by Krzysiek 20 May '08

20 May '08

Hi there Is it possible in Freeswitch to disable default profile on port 5060 and leave only tls profile on port 5061, so Freeswitch won't accept registrations other then through tls? If i'm tring to do something stupid, feel free to correct me :) Now I have something like this: "sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================ outbound profile sip:mod_sofia@192.168.1.2:5080 RUNNING (0) nat profile sip:mod_sofia@192.168.1.2:5070 RUNNING (0) default profile sip:mod_sofia@192.168.1.2:5060 RUNNING (0) default profile sip:mod_sofia@192.168.1.2:5061 RUNNING (0) (TLS) 192.168.1.2 alias default ALIASED ================================================================================================= 3 profiles 1 alias " TIA Chris

3 4

wanpipe compiling problem
by Helmut Kuper 20 May '08

20 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get wanpipe module compiled. I don't want to use that libpri-1.2.6 and wanpipe-3.1.1 but my already installed versions of them. ./configure allows me to do this via --with-libpri and - --with-wanpipe dir options. (./configure --prefix=/opt/app/freeswitch -C - --with-libpri=/usr/src/libpri-1.4.1 --with-wanpipe=/usr/src/wanpipe-3.1.4) Unfortunately the Makefile in mod_wanpipe still tries to download libpri-1.2.6 (which isn't available on http://downloads.digium.com/pub/libpri) as well as wanpipe-3.1.1, which is not neccessary. Is there a way to disable the downloads and compilings of libpri and wanpipe during compiling freeswitch? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgywOEACgkQ4tZeNddg3dzueACgmXcayPoH/ksIjwFysThMXg4/ wmEAn1q9QqKgoM27mLFFA9rgcY+9UNlm =evsT -----END PGP SIGNATURE-----

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How to reject a call
by Czaderna 20 May '08

20 May '08

Hello I've got question about rejecting a calls. I make a call between two wireless phones (not mobile phones). When first phone starts ringing I want to reject this call but I can't because feeswitch still trying ring to me. (and traying, trying, trying, ...) On the console I writing something like this: originate {ignore_early_media=true,bypass_media=true,continue_on_fail=false,hangup_after_bridge=true}sofia/gateway/voipdiscount.com/0048xxxxxxxx1 &bridge(sofia/gateway/voipdiscount.com/0048xxxxxxxx2) Is there any variable which I should set up to reject a call or I should do something else ? Thanks.

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att_xfer application
by Csaba Zelei 20 May '08

20 May '08

Hi, Currently I need to set up attended transfer to phones which dont have any support to it. So, made a simple extension in features.xml which read the number and supply it to att_xfer app: <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="read" data="3 4 sounds/mellekek.wav attxfer_callthis 30000 #"/> <action application="att_xfer" data="sofia/default/${attxfer_callthis}"/> </condition> </extension> then bind it to dtmf 3 in default.xml Local Extension with: <anti-action application="bind_meta_app" data="3 a a execute_extension::att_xfer XML features"/> I can activate this feautre, it's hold the bleg and call the 3rd party, but when I try to make the transfer with dtmf 0 than it hangup all legs. For me it's look like that FS hangup the channels than try to bridge them. Here is the console log: freeswitch@devil4> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate 19112de8-2197-11dd-ba14-5dc4d1b7fe8e,2008-05-14 11:21:13,1210756873,sofia/default/anonymous@localhost,CS_EXECUTE,1003,1003,192.168.1.192,1001,att_xfer,sofia/default/1000,XML,default,PCMA,8000,PCMA,8000 193e7bf4-2197-11dd-ba14-5dc4d1b7fe8e,2008-05-14 11:21:13,1210756873,sofia/default/1001,CS_EXCHANGE_MEDIA,1003,1003,192.168.1.192,1001,playback,local_stream://moh,XML,default,PCMA,8000,PCMA,8000 1e67e516-2197-11dd-ba14-5dc4d1b7fe8e,2008-05-14 11:21:22,1210756882,sofia/default/1000,CS_EXCHANGE_MEDIA,1003,1003,192.168.1.192,1000,,,XML,default,PCMA,8000,PCMA,8000 3 total. freeswitch@devil4> 2008-05-14 11:21:40 [DEBUG] mod_dptools.c:1344 xfer_on_dtmf() sofia/default/1000 State Change CS_EXCHANGE_MEDIA -> CS_EXECUTE 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:671 switch_core_session_signal_state_change() Kill sofia/default/1000 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:193 audio_bridge_thread() sofia/default/anonymous@localhost ended call via DTMF 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:194 audio_bridge_thread() Kill sofia/default/1000 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/anonymous@localhost [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/default/1000 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/anonymous@localhost] 2008-05-14 11:21:40 [NOTICE] switch_ivr_bridge.c:802 switch_ivr_multi_threaded_bridge() Hangup sofia/default/anonymous@localhost [CS_EXECUTE] [UNALLOCATED] 2008-05-14 11:21:40 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/anonymous@localhost [KILL] 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:671 switch_core_session_signal_state_change() Kill sofia/default/anonymous@localhost [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/1000 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/default/anonymous@localhost [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/1000] 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/default/1000) State EXCHANGE_MEDIA going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1000 Running State Change CS_EXECUTE 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/default/1000) State EXECUTE 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:156 sofia_on_execute() sofia/default/1000 SOFIA EXECUTE 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:128 switch_core_standard_on_execute() Standard EXECUTE 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:139 switch_core_standard_on_execute() sofia/default/1000 Execute three_way(19112de8-2197-11dd-ba14-5dc4d1b7fe8e) 2008-05-14 11:21:40 [DEBUG] switch_core_media_bug.c:227 switch_core_media_bug_add() Attaching BUG to sofia/default/anonymous@localhost 2008-05-14 11:21:40 [DEBUG] switch_core_media_bug.c:312 switch_core_media_bug_close() Removing BUG from sofia/default/anonymous@localhost 2008-05-14 11:21:40 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/default/1000 [CS_EXECUTE] [NORMAL_CLEARING] 2008-05-14 11:21:40 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/1000 [KILL] 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:671 switch_core_session_signal_state_change() Kill sofia/default/1000 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/default/1000) State EXECUTE going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1000 Running State Change CS_HANGUP 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/1000) State HANGUP 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/default/1000 hanging up, cause: NORMAL_CLEARING 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:296 sofia_on_hangup() Sending BYE to sofia/default/1000 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/1000, cause: NORMAL_CLEARING 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/1000) State HANGUP going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:731 switch_core_session_thread() Session 5 (sofia/default/1000) Locked, Waiting on external entities 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:749 switch_core_session_thread() Session 5 (sofia/default/1000) Ended 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:751 switch_core_session_thread() Close Channel sofia/default/1000 [CS_HANGUP] 2008-05-14 11:21:40 [DEBUG] switch_ivr_play_say.c:1130 switch_ivr_play_file() done playing file 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/1001 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:160 audio_bridge_thread() Kill sofia/default/anonymous@localhost [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:383 switch_core_session_receive_message() Kill sofia/default/1001 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/default/anonymous@localhost [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/1001] 2008-05-14 11:21:40 [NOTICE] switch_ivr_bridge.c:374 audio_bridge_on_exchange_media() Hangup sofia/default/1001 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2008-05-14 11:21:40 [DEBUG] switch_channel.c:1355 switch_channel_perform_hangup() Kill sofia/default/1001 [KILL] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:919 switch_ivr_uuid_bridge() originator uuid 1e67e516-2197-11dd-ba14-5dc4d1b7fe8e is not present 2008-05-14 11:21:40 [DEBUG] mod_dptools.c:1371 hanguphook() Hangup Command uuid_bridge(1e67e516-2197-11dd-ba14-5dc4d1b7fe8e 19112de8-2197-11dd-ba14-5dc4d1b7fe8e): -ERR Invalid uuid 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:671 switch_core_session_signal_state_change() Kill sofia/default/1001 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:431 switch_core_session_run() (sofia/default/1001) State EXCHANGE_MEDIA going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/1001 Running State Change CS_HANGUP 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/1001) State HANGUP 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/default/1001 hanging up, cause: NORMAL_CLEARING 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:296 sofia_on_hangup() Sending BYE to sofia/default/1001 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/1001, cause: NORMAL_CLEARING 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/1001) State HANGUP going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:731 switch_core_session_thread() Session 4 (sofia/default/1001) Locked, Waiting on external entities 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:160 audio_bridge_thread() Kill sofia/default/1001 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:275 audio_bridge_thread() read: sofia/default/anonymous@localhost Bad Frame.... Bubye! 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:344 audio_bridge_thread() Kill sofia/default/1001 [BREAK] 2008-05-14 11:21:40 [DEBUG] switch_ivr_bridge.c:347 audio_bridge_thread() BRIDGE THREAD DONE [sofia/default/anonymous@localhost] 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:428 switch_core_session_run() (sofia/default/anonymous@localhost) State EXECUTE going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:364 switch_core_session_run() sofia/default/anonymous@localhost Running State Change CS_HANGUP 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/anonymous@localhost) State HANGUP 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/default/anonymous@localhost hanging up, cause: UNALLOCATED 2008-05-14 11:21:40 [DEBUG] mod_sofia.c:296 sofia_on_hangup() Sending BYE to sofia/default/anonymous@localhost 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/default/anonymous@localhost, cause: UNALLOCATED 2008-05-14 11:21:40 [DEBUG] switch_core_state_machine.c:392 switch_core_session_run() (sofia/default/anonymous@localhost) State HANGUP going to sleep 2008-05-14 11:21:40 [DEBUG] switch_core_session.c:731 switch_core_session_thread() Session 3 (sofia/default/anonymous@localhost) Locked, Waiting on external entities 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:749 switch_core_session_thread() Session 3 (sofia/default/anonymous@localhost) Ended 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:751 switch_core_session_thread() Close Channel sofia/default/anonymous@localhost [CS_HANGUP] 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:749 switch_core_session_thread() Session 4 (sofia/default/1001) Ended 2008-05-14 11:21:40 [NOTICE] switch_core_session.c:751 switch_core_session_thread() Close Channel sofia/default/100 My question is if its a bug or I did something wrong? I used the lastest trunk on Ubuntu linux. Thanks, Csaba

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Problem during shutdown?
by Helmut Kuper 20 May '08

20 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just tried to shutdown freeswitch from console. I found that FS sometimes hangs for a couple of seconds at always the same point. Unfortunately the timestamp of each log line doesn't reflect this. console log: 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1068 do_shutdown() Stopping: mod_console 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1079 do_shutdown() mod_console shutdown. 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1068 do_shutdown() Stopping: mod_logfile 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1079 do_shutdown() mod_logfile shutdown. 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1068 do_shutdown() Stopping: mod_sofia 2008-05-19 16:52:41 [NOTICE] switch_loadable_module.c:503 switch_loadable_module_unprocess() Deleting API Function 'conference' 2008-05-19 16:52:41 [CONSOLE] sofia_presence.c:591 sofia_presence_event_thread_run() Event Thread Ended ... hanging for 20-30 seconds ... 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1079 do_shutdown() mod_sofia shutdown. 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1068 do_shutdown() Stopping: mod_spidermonkey 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1079 do_shutdown() mod_spidermonkey shutdown. 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1068 do_shutdown() Stopping: mod_enum 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1079 do_shutdown() mod_enum shutdown. 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_fsv has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_g729 has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_cdr_csv has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_h26x has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_dialplan_xml has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_voipcodecs has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_dialplan_asterisk has no shutdown routine 2008-05-19 16:52:41 [CONSOLE] switch_loadable_module.c:1082 do_shutdown() mod_ilbc has no shutdown r regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgxmhAACgkQ4tZeNddg3dz4PwCaAjPL4aLslWSWN/qjH8KyvelW 94gAnidZZuciZ9U9vCe9Tt6eTek10L0d =RGHQ -----END PGP SIGNATURE-----

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This weeks meeting agenda!
by Brian West 19 May '08

19 May '08

http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_21 This week is a bit light and more will be added before wednesday rolls around. On to a small reminder: This is a requirement if you're following trunk: svn update make sofia-reconf current vm-sync Please issue that command if you encounter any strange issues or crashes. This should take care of it. Thanks, Brian West sip:brian@freeswitch.org 1000*543 (freenum)

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Snom360 softphone and Freeswitch
by Krzysiek 19 May '08

19 May '08

Hi I have recently tested snom 360 softphone (www.snom.com/download/snom360-5.3.exe) with Freeswitch. It has SRTP* and TLS support. It is quite old (2006) and I'm wondering if this softphone creates sip sessions (with tls enabled) properly. Here is a sip trace (INVITE message should be sufficient) : ------------------------------------------------------------------------ recv 1092 bytes from tls/[192.168.1.4]:1145 at 17:09:19.945705: ------------------------------------------------------------------------ INVITE sip:1002@192.168.1.2;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.4:1145;branch=z9hG4bK-k1yp4ytettb1;rport From: "1001" <sip:1001@192.168.1.2>;tag=9udth2g3o3 To: <sip:1002@192.168.1.2;user=phone> Call-ID: 50b43148ee7f-696c59fq7vjl@snomSoft-000413FFFFFF CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:1001@192.168.1.4:1145;transport=tls;line=ojn9itpa>;flow-id=1 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snomSoft/5.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Content-Type: application/sdp Content-Length: 362 v=0 o=root 28625 28625 IN IP4 192.168.1.4 s=call c=IN IP4 192.168.1.4 t=0 0 m=audio 57428 RTP/AVP 0 8 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:IZpDi51+JeVKPOO3Mox0q3jZYmJorsThpl6b2jw1 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv I have read in rfc3261 that : "The use of "transport=tls" has consequently been deprecated" So does snom 360 create sessions properly or not? Freeswitch works with it quite good but I don't know if it is right according to rfc. I tested few free/opensource softphones such as Minisip, Lynxphone, PhonerLite and neither worked properly with freeswitch with srtp/tls enabled. PhonerLite crashed when the call was to be setup although it registered properly , Minisip created tls sip sessions in weird way (transport=tcp) that the second leg of the call wasn't created (I wrote about this here http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003179.html) and Lynxphone didn't even register. So snom will be first FREE softphone that works. *I know that this softphone has bad SRTP/SDES implementation (RTP/AVP instead of RTP/SAVP) but I think that it isn't such a big problem when media streams by-pass Freeswitch server (for example when calls are setup in local LAN and Inbound-no-media is set to true). Am I right? TIA for reply Chris PS Sorry for my english.

2 1

Need help with Openzap and Sangoma A200 analog card
by George Howitt 19 May '08

19 May '08

Hello all, I have a Sangoma A200 card with 2 FXS and 2 FXO interfaces. Zaptel 1.4.10 .1 is installed, along with wanpipe-3.2.5. FreeSwitch is compiled and installed with OpenZap. However, I have no idea how to configure OpenZap for my card. I can't find any examples. Specifically, how do I configure /etc/openzap/openzap.conf ???

4 3

TLS-Support
by Helmut Kuper 19 May '08

19 May '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I try to get SIPS via TLS/TCP up. Unfortunately I get this error in console as a non root user: tport_bind_server(0x80febb8) to */85.16.246.5:5061/sips nta: bind(85.16.246.5:5061;transport=*): Protocol not supported nua: initializing SIP stack failed tport_destroy(0x80febb8) 2008-05-19 14:27:06 [ERR] sofia.c:551 sofia_profile_thread_run() Error Creating SIP UA for profile: internal in sip_profiles/internal.xml I configured TLS support as follows: ~  ~ <param name="tls" value="true"/> ~  ~ <param name="tls-bind-params" value="transport=tls"/> ~  ~ <param name="tls-sip-port" value="5061"/> ~  ~ <param name="tls-cert-dir" value="$${base_dir}/conf.tls/ssl"/> ~  ~ <param name="tls-version" value="tlsv1"/> I did a "bin/gentls_cert setup" to generate a CA. Maybe I habe to compile FS with TLS support? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkgxeBoACgkQ4tZeNddg3dyCpwCgj7pfQvJKMWKevJVyeIU6dQmy xcwAnidYgk6rA6p0YziRHkbJ1biQYuBA =DIYB -----END PGP SIGNATURE-----

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