September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_xml_curl - sample php example file location
by Aloysius Lloyd 23 Sep '10

23 Sep '10

Hi All, Help me find the Example directory.php and dialplan.php for the mod_xml_curl. Wiki says "There is a PHP example that does everything from a webserver, with per switch configuration capability in FreeSWITCH™ SVN ( scripts/contrib/trixter/xml-curl<http://fisheye.freeswitch.org/browse/Unsupported/trixter/xml-curl> )." But the link is broken. Any help is appreciated. Thanks Lloyd

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Re: [Freeswitch-users] sip stack problem in fsw
by JRichey 23 Sep '10

23 Sep '10

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

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Sending Alpha Numeric characters using uuid_send_dtmf
by rex.alex 23 Sep '10

23 Sep '10

Hello, I am trying to send alpha numeric characters when responding to an IVR like, "api uuid_send_dtmf 6a677f06-fa1a-4257-b566-32fe2715b3dc WMW081". But it is not recognizing whereas the numeric characters are getting recognized. Please assist, how do I send? Thanks, Rex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sending-Alpha-Numeric-charact… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Freeswitch Register Failure
by Raja Rokkam 23 Sep '10

23 Sep '10

Hi All, I have setup FreeSWITCH Version 1.0.6 (hacked-20100922T074940Z) on a Ubuntu 10.0.4 Virtual Machine on my Windows XP desktop. I am using another softphone X-Lite to connect to it. The VM uses a bridged network so essentially has its own IP assigned by a DHCP server. For testing I am using 1000.xml and changed the password fields to 1234. However I am not able to register. The error I get is: 2010-09-23 11:55:48.240027 [WARNING] sofia_reg.c:1030 SIP auth failure (REGISTER) on sofia profile 'internal' for [1000(a)192.168.6.210] from ip 192.168.6.144 Can anyone please let me know what other configurations I have to make ? Thanks, Raja.

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Codec Error! v=0
by Frank Ochere 23 Sep '10

23 Sep '10

Hello, I've been seeing the error below for calls sent to a particular SIP gateway, anyone knows what it means and should I be concerned? (I use g729 for my inbound and outbound codec prefs in FS) 2010-09-22 10:34:23.398832 [ERR] sofia.c:4919 Codec Error! v=0 o=00345254725065043 1285213687 1285213687 IN IP4 1.1.1.1 s=AddPac Gateway SDP c=IN IP4 1.1.1.1 t=1285213687 0 m=audio 24686 RTP/AVP 18 a=rtpmap:18 G729/8000/1 a=recvonly a=ptime:20 Kind Regards Frank Ochere

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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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