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Problem starting internal profile (pfSense - freeswitch)
by bsd 22 Sep '10

22 Sep '10

Hello, I am actually trying to use the FreeSwitch as provided by pfSense for version 1.2.3 So far I have been able to start • "profile external" which binds to the WAN address of my pfSense Box • "profile internal-ipv6" which binds to the ipv6 LAN address of my pfSense Box But profile "internal" does not want to start… When I have a look at the debug I have : > 2010-09-22 12:18:34.192432 [NOTICE] sofia.c:1508 Adding Alias [82.xx.tt.242] for profile [internal] > 2010-09-22 12:18:34.192432 [NOTICE] sofia.c:2841 Started Profile internal [sofia_reg_internal] > 2010-09-22 12:18:34.192432 [DEBUG] sofia.c:812 Creating agent for internal > tport_bind_server(0x28e6b540) to */82.xx.tt.242:5060/sip > tport_alloc_primary(0x28e6b540): new primary tport 0x28e72880 > nta: bind(82.xx.tt.242:5060;transport=*): Address already in use > nua: initializing SIP stack failed > 2010-09-22 12:18:34.267813 [ERR] sofia.c:862 Error Creating SIP UA for profile: internal Do you know what causes this error… ?? From what I can read / understand, It seems that port 5060 is already in use… ?? netstat, tells me that : # netstat -an Active Internet connections (including servers) Proto Recv-Q Send-Q Local Address Foreign Address (state) tcp4 0 52 82.xx.tt.242.22 202.30.198.57.36032 ESTABLISHED tcp4 0 0 192.168.2.1.22 192.168.2.21.55899 ESTABLISHED tcp4 0 0 82.xx.tt.242.5080 *.* LISTEN tcp4 0 0 *.8021 *.* LISTEN tcp6 0 0 ::1.5060 *.* LISTEN tcp4 0 0 *.10050 *.* LISTEN tcp4 0 0 127.0.0.1.80 *.* LISTEN tcp4 0 0 192.168.2.1.8080 *.* LISTEN tcp6 0 0 *.53 *.* LISTEN tcp4 0 0 *.53 *.* LISTEN tcp4 0 0 82.xx.tt.242.1195 82.229.142.27.16948 ESTABLISHED tcp4 0 0 *.2189 *.* LISTEN tcp4 0 0 *.443 *.* LISTEN tcp4 0 0 127.0.0.1.8021 *.* LISTEN tcp4 0 0 *.1194 *.* LISTEN tcp4 0 0 *.22 *.* LISTEN tcp6 0 0 *.22 *.* LISTEN udp4 0 0 82.xx.tt.242.30439 195.220.94.163.123 udp4 0 0 82.xx.tt.242.62089 81.93.183.116.123 udp4 0 0 82.xx.tt.242.14264 87.98.206.99.53 udp4 0 0 82.xx.tt.242.54807 87.98.206.98.53 udp4 0 0 82.xx.tt.242.5080 *.* udp6 0 0 ::1.5060 *.* udp4 0 0 82.xx.tt.242.24868 82.66.93.254.5351 udp4 0 0 *.3401 *.* udp4 0 0 *.4827 *.* udp4 0 0 *.35361 *.* udp4 0 0 *.514 *.* udp6 0 0 *.514 *.* udp4 0 0 82.xx.tt.242.32376 91.121.79.182.123 udp4 0 0 82.xx.tt.242.14724 91.121.120.27.123 udp4 0 0 *.67 *.* udp6 0 0 *.53 *.* udp4 0 0 *.53 *.* udp4 0 0 *.28008 *.* udp4 0 0 *.37049 *.* udp4 0 0 *.5353 *.* udp4 0 0 *.5060 *.* udp4 0 0 192.168.2.1.50235 *.* udp4 0 0 *.1900 *.* udp4 0 0 *.161 *.* udp4 0 0 *.* *.* udp4 0 0 *.* *.* icm4 0 0 *.* *.* Any idea how to debug that… ? ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ Gregober ---> PGP ID --> 0x1BA3C2FD bsd @at@ todoo.biz ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

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Matching plus sign in dialplan Regexp
by bsd 22 Sep '10

22 Sep '10

Hello folks, I am actually using a regexp such as ^(\d{7,15})$ to route call to one of my gateway. My SIP clients have all their number registered with number such as +33123456789 Is there a way to treat the "+" and transform It in "00" at the level of FreeSwitch (I can do that at the client level, but you'll understand that It is easier to do on the server side). Thanks. ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ Gregober ---> PGP ID --> 0x1BA3C2FD bsd @at@ todoo.biz ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

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Weekly Conference Call - Starts at 1pm EST
by Norman Tomlins 22 Sep '10

22 Sep '10

Hi All, The weekly conference call starts in 15 minutes. You can call in using either PSTN at +1-919-386-9900 SIP: 888(a)conference.freeswitch.org GTALK: conf+888(a)conference.freeswitch.org<conf%2B888(a)conference.freeswitch.org> This week topic is mod_xml_curl Norman Tomlins Voice Network Inc. http://www.VoiceNetwork.ca

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Question on SIP INFO
by DJB International 22 Sep '10

22 Sep '10

When I tried to set the variable dtmf_type=info, or in sip profile parameter dtmf-type=info, would FS send the SIP INFO packet. For instance, if I called an IVR and pressed 2 after the call got answered by the IVR, I was expecting to see the message similar to below, but I did not see it. Is there any other variables that needs to be set? INFO sip:2143302100@172.17.2.33 <sip%3A2143302100(a)172.17.2.33> SIP/2.0 Via: SIP/2.0/UDP 172.80.2.100:5060 From: <sip:9724401003@172.80.2.100 <sip%3A9724401003(a)172.80.2.100>>;tag=43 To: <sip:2143302100@172.17.2.33 <sip%3A2143302100(a)172.17.2.33> >;tag=9753.0207 Call-ID: 984072_15401962(a)172.80.2.100 CSeq: 25634 INFO Supported: 100rel Supported: timer Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2 Duration= 215 Thank you.

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TLS re-negotiation attack on SIP/TLS of FreeSWITCH?
by Fabio Pietrosanti (naif) 22 Sep '10

22 Sep '10

Hi all, i read about the TLS-RENEGOTIATION vulnerability: http://www.educatedguesswork.org/2009/11/understanding_the_tls_renegoti.html http://www.sslshopper.com/article-ssl-and-tls-renegotiation-vulnerability-d… www.phonefactor.com/sslgapdocs/Renegotiating_TLS.pdf Does the FreeSWITCH SIP/TLS implementation suffer from the TLS Renegotiation vulnerability or the TLS-renegotiation it's disabled by default, in how OpenSSL is used? Fabio Pietrosanti

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files.freeswitch.org
by Brian West 22 Sep '10

22 Sep '10

YES we know files.freeswitch.org is down, hardware failure Dell will be on site to fix the motherboard the data is fine the nic on the MB failed some how. /b

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FS terminate when trying to load mod_h323
by gamalsa 21 Sep '10

21 Sep '10

Mod_h323 compiled and build successfully but when FS trying to load, it terminates with the following error: terminate called after throwing an instance of '__gnu_cxx::__concurrence_unlock_error' what(): __gnu_cxx::__concurrence_unlock_error -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-terminate-when-trying-to-l… Sent from the freeswitch-users mailing list archive at Nabble.com.

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all sip registrations intermittently failing
by covici＠ccs.covici.com 21 Sep '10

21 Sep '10

Hi. Suddenly all sip registrations to sip providers are failing -- it says its for ping, but manual pings always succeed and so I am quite confused. I get warnings about failed registrations, but sometimes they are up, so I have no idea as to what is happening. Not sure how to debug this one. Should I get rid of the ping parms in the gateways? This all has worked before, but I have done make current since, so maybe something broke? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Re: [Freeswitch-users] sip stack problem in fsw
by JRichey 21 Sep '10

21 Sep '10

It appears to be the lack of a continuous stream of RTP packets that is the issue, which I've incorrectly referred to as VAD. Is there a way to cause FreeSwitch to send a continuous stream of RTP packets even when there's really nothing to send? -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Anthony Minessale Sent: Tuesday, September 21, 2010 9:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] sip stack problem in fsw are you in a conference running on FreeSWITCH or just bridging a phone out to a far end? There is nothing in FreeSWITCH that runs vad by default. There are vad params in the RTP stack but they are disabled by default      <!-- If these are not set then its the actual phone who is choosing to do vad not FreeSWITCH so disable it in your phone. any problems you are having in this regard are 100% to do with who you are calling being only partially implemented to carry voice over ip. On Mon, Sep 20, 2010 at 5:34 PM, JRichey <jrichey(a)itltd.net> wrote: > The gateway sending calls to the FreeSwitch box ends up dropping the calls. > There are also some pops and clicks that aren't there when playing a > continuous audio file. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Brian > West > Sent: Monday, September 20, 2010 12:23 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] sip stack problem in fsw > > > You shouldn't see RTP between files when sleeping... its 100% valid to NOT > send any media when their is NOTHING to be sent. Does your phone freak out > or something? > > /b > > On Sep 20, 2010, at 2:08 PM, JRichey wrote: > >> I'm in a similar situation where I need to disable VAD because the other > end >> will disconnect the call if the RTP stops. I've tried the setting Anthony >> suggested as well as several others I've found on the wiki, but I've not >> been able to get it to work. The variables I've been trying are listed >> below. >> >> suppress-cng >> suppress_cng >> rtp_disable_vad_out >> rtp_disable_vad_in >> bridge_generate_comfort_noise >> send_silence_when_idle >> >> >> For my test calls I'm just playing back sound files with sleep in-between >> them and I see no RTP when calling sleep. I'm using tcpdump and wireshark >> to analyze the calls. >> >> >> -Justin Richey >> jrichey(a)ITLtd.net > > > _______________________________________________ > 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 > > > > _______________________________________________ > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _______________________________________________ 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

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Compile Error
by Nathaniel L Keeling 21 Sep '10

21 Sep '10

I am trying to compile FreeSwitch on a Solaris 10 sparc machine and I am getting this error: Compiling src/switch_core_sqldb.c ... Compiling src/switch_core_session.c ... src/switch_core_session.c: In function `switch_core_session_thread': src/switch_core_session.c:1193: warning: long int format, switch_size_t arg (arg 8) src/switch_core_session.c:1203: warning: long int format, switch_size_t arg (arg 8) src/switch_core_session.c:1221: warning: long int format, switch_size_t arg (arg 8) gmake[1]: *** [libfreeswitch_la-switch_core_session.lo] Error 1 gmake: *** [all] Error 2 I looked in the source directory and I was able to see this file; "libfreeswitch_la-switch_core_session.loT" Any help would be appreciated. Thanks Nathaniel

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