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Re: [Freeswitch-users] Hacker Attack?
by Peter Olsson 01 Feb '11

01 Feb '11

iptables is a Linux command. /Peter ----- Reply message ----- Från: "Joao Leme" <joaocarlosleme(a)gmail.com> Datum: sön, jan 30, 2011 13:56 Rubrik: [SPAM] - Re: [Freeswitch-users] Hacker Attack? Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> I tried "iptables -I INPUT -s [212.224.71.236] -j DROP" and got " Unknown command: iptables...". Do I must install fail2ban to issue iptables command? I'm on windows 7. Thanks On Sat, Jan 29, 2011 at 4:26 PM, curriegrad2004 <curriegrad2004(a)gmail.com<mailto:curriegrad2004@gmail.com>> wrote: iptables -I INPUT -s [hackerip] -j DROP A better solution is searching the wiki for fail2ban with FreeSwitch. On Sat, Jan 29, 2011 at 4:20 PM, Joao Leme <joaocarlosleme(a)gmail.com<mailto:joaocarlosleme@gmail.com>> wrote: > How do I do that? > Thanks! > On Sat, Jan 29, 2011 at 4:12 PM, curriegrad2004 <curriegrad2004(a)gmail.com<mailto:curriegrad2004@gmail.com>> > wrote: >> >> Try using iptables and block all incoming traffic from this specific host? >> >> On Sat, Jan 29, 2011 at 3:39 PM, Joao Leme <joaocarlosleme(a)gmail.com<mailto:joaocarlosleme@gmail.com>> >> wrote: >> > I just downloaded and compiled the latest Git and a little after >> > starting >> > freeswitch I'm getting non stop the following: >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [140(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > [WARNING] sofia_reg.c:1247 SIP auth challenge (REGISTER) on sofia >> > profile >> > ‘internal’ for [thomas(a)76.XXX.XX.XXX] from ip 212.224.71.236 >> > it's non-stop and doesn't let me do nothing else. After the first time I >> > went on to vars and changed the 1234 password....restarted and same >> > thing >> > happened, I also try denying the ip on acl.conf (not sure if has >> > something >> > to do with it but gave it a try): >> > >> > <configuration name="acl.conf" description="Network Lists"> >> > <network-lists> >> > <list name="test2" default="allow"> >> > <node type="deny" host="212.224.71.236" >> > mask="255.255.255.0"/> >> > </list> >> > </network-lists> >> > </configuration> >> > >> > Restarted the computer but nothing, he (thomas I guess) was back on my >> > console. >> > >> > Any ideas??? p.s. My computer is on DMZ (I know DMZ is not ideal but is >> > the >> > only way I got to be able to connect to the internal profile from out of >> > the >> > office etc). >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@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<mailto:FreeSWITCH-users@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<mailto:FreeSWITCH-users@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4d450b3232767678720833!

4 3

Blind Transfer Not working
by lakshmanan ganapathy 01 Feb '11

01 Feb '11

Hi all, here is my dialplan <context name="default"> <extension name="outbound_soc"> <condition field="destination_number" expression="^.*$"> <action application="log" data="ERR ${domain_name}"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="bypass_media=false"/> <action application="ring_ready"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="exec_after_bridge_app=park"/> <action application="export" data="RECORD_STEREO=true"/> <action application="bridge" data="user/1000"/> </condition> </extension> </context> I made a call from 9952248266 to 39114601. As expected it called to the 1000 extension and I answered the call. I was using twinkle as a softphone. Now from twinkle I initiated "Transfer". I've given "9976975781" as a number and I choose "Blind Transfer". But the call didn't get proceeded. Here is the log with sofia trace enabled. http://pastebin.freeswitch.org/15189 Once REFER is received by freeswitch, It replied with "Accept" for that refer, but after that the call was not proceeding!!

1 1

Question about DTMF duration
by Yehavi Bourvine 01 Feb '11

01 Feb '11

Hello, While trying to debug DTMF problems with some gateways I came across the following issue: The phone generates the DTMF of a certain period, but I would like the PBX to send a longer period to the gateway. Thus, I change switch.conf.xml to have *min-dtmf-duration=*80000 and at the outgoing profile toward the gateway I put *dtmf-duration=*80000; I tried large values in order to see the difference. However, wireshark stil shows much smaller duration (around 1,000). Do I understand something incorrectly? Thanks, __Yehavi:

1 0

mod_callcenter and effective_caller_id_name
by Weigel, Stefan 01 Feb '11

01 Feb '11

Hi all, when modifying 'effective_caller_id_name' it's working for user extensions. When routing to application mod_callcenter the change doesn't take effect. Is it a bug or am I missing something ? Thanks and best regards Stefan

1 1

How to play Background music during conference?
by kapil.rastogi 01 Feb '11

01 Feb '11

Dear Friends, I am using javascript as a scripting language in freeSWITCH. I want to play the background music during conference without mute any member in the conference room. I tried "perpetual-sound" but it is not working. Can anyone help me about this? If yes, then please also send me the example code to play background music. ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-play-Background-music-… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 2

How to play wav files from other path?
by kapil.rastogi 01 Feb '11

01 Feb '11

Hi, I want to play wav files from the different path. As now it is playing the wav files from: "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/" path. Please tell me how to play wav files from other path using javascript application. Thanks in advance.... ----- Regards, Kapil Rastogi Telemune Software Solutions P Ltd. kapil.rastogi(a)telemune.net +919013204760 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-play-wav-files-from-ot… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 2

Remote LogIn to Freeswitch?
by Joao Leme 01 Feb '11

01 Feb '11

Hi There, What do I have to do to be able to LogIn to Freeswitch from Home (server is located at office) starting from the basic/original configuration? I'm using X-Lite. I've been able to LogIn replacing the internal IP by the external IP from the Office but the sound is not working so I wanted to know what are the configuration changes that have to be done to allow it. Do I have to create a different profile? I want be able to do the same just as if I was at the Office. Thanks, John

5 9

sip_contact_host
by Madovsky 01 Feb '11

01 Feb '11

I set my internal profile with a domain but sip_contact_host stays an IP. how to change it ? Thanks

2 3

Outbound only calls don't connect when bypass_media is true.
by Marcin Wojtowicz 31 Jan '11

31 Jan '11

Hello, I'm a new user of freeswitch, so please bear with me. I have the following setup: voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP -> my nokia cellphone on AT&T wireless. This setup is intended to conserve the battery usage. I've managed to make everything work well when I'm calling in over any phone to my cell phone, and freeswitch is enabled to work in bypass_media = true, even though by cell is behind NAT on at&t's network. Things break when I pick up my cell and try to call my home phone (or any phone for that matter). This is the relevant snippet from my dialplan: <extension name="outbound"> <condition field="destination_number" expression="^1?([2-9]\d{2}[2-9]\d{6})$">  <action application="bridge" data="sofia/gateway/voip.ms/1$1"/> </condition> </extension> Like shown above, my call will go to my home phone. When I uncomment the bypass_media tag, my call will not connect. Here are the siptraces I replaced my real home phone number in the with "MYPHONE". recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250: ------------------------------------------------------------------------ INVITE sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100> Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP> Supported: 100rel,timer CSeq: 5244503 INVITE Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE User-Agent: S60 RM-624 v 20.2.042 (en) Expires: 120 Privacy: None Session-Expires: 1800 Max-Forwards: 70 Content-Type: application/sdp Accept-Language: en Content-Length: 292 v=0 o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6 s=- c=IN IP4 10.153.174.6 t=0 0 m=audio 49152 RTP/AVP 18 97 98 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=maxptime:40 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 ------------------------------------------------------------------------ send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180 From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100> Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244503 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Content-Length: 0 ------------------------------------------------------------------------ send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180 From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on sofia profile 'internal' for [MYPHONE(a)192.168.1.100] from ip 32.136.78.180 To: <sip:MYPHONE@192.168.1.100>;tag=FQy5v5emcyt1m Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244503 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="192.168.1.100", nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625: ------------------------------------------------------------------------ ACK sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100>;tag=FQy5v5emcyt1m Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244503 ACK Supported: sec-agree Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250: ------------------------------------------------------------------------ INVITE sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100> Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP> Supported: 100rel,timer CSeq: 5244504 INVITE Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE User-Agent: S60 RM-624 v 20.2.042 (en) Expires: 120 Privacy: None Session-Expires: 1800 Max-Forwards: 70 Proxy-Authorization: Digest qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE@192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913" Content-Type: application/sdp Accept-Language: en Content-Length: 292 v=0 o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6 s=- c=IN IP4 10.153.174.6 t=0 0 m=audio 49152 RTP/AVP 18 97 98 a=sendrecv a=rtpmap:18 G729/8000 a=ptime:20 a=maxptime:40 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 ------------------------------------------------------------------------ send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180 From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100> Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244504 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Content-Length: 0 ------------------------------------------------------------------------ 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel sofia/internal/1001(a)192.168.1.100 [e5841001-04bd-4e16-9519-64ff2c7a8c2f] 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001 <1001>->MYPHONE in context default 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0] send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125: ------------------------------------------------------------------------ INVITE sip:1MYPHONE@newyork.voip.ms SIP/2.0 Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS Max-Forwards: 69 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms> Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788615 INVITE Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 280 X-FS-Support: update_display Remote-Party-ID: "Extension 1001" <sip:1001@69.125.20.15>;party=calling;screen=yes;privacy=off v=0 o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6 s=- c=IN IP4 10.153.174.6 t=0 0 m=audio 49152 RTP/AVP 18 97 98 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 a=ptime:20 a=maxptime:40 ------------------------------------------------------------------------ recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms>;tag=as7e7ea843 Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788615 INVITE User-Agent: VoIPMS/SERAST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms", nonce="2d534dd6" Content-Length: 0 ------------------------------------------------------------------------ send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750: ------------------------------------------------------------------------ ACK sip:1MYPHONE@newyork.voip.ms SIP/2.0 Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS Max-Forwards: 69 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms>;tag=as7e7ea843 Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788615 ACK Content-Length: 0 ------------------------------------------------------------------------ send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750: ------------------------------------------------------------------------ INVITE sip:1MYPHONE@newyork.voip.ms SIP/2.0 Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN Max-Forwards: 69 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms> Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788616 INVITE Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms> Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Proxy-Authorization: Digest username="121628", realm="newyork.voip.ms", nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE@newyork.voip.ms", response="16f3301efae13df926da7550f709d28a" Content-Type: application/sdp Content-Disposition: session Content-Length: 280 X-FS-Support: update_display Remote-Party-ID: "Extension 1001" <sip:1001@69.125.20.15>;party=calling;screen=yes;privacy=off v=0 o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6 s=- c=IN IP4 10.153.174.6 t=0 0 m=audio 49152 RTP/AVP 18 97 98 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:98 telephone-event/8000 a=fmtp:98 0-15 a=ptime:20 a=maxptime:40 ------------------------------------------------------------------------ recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms> Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788616 INVITE User-Agent: VoIPMS/SERAST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:1MYPHONE@74.63.41.218> Content-Length: 0 ------------------------------------------------------------------------ recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500: ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms>;tag=as632cb7d9 Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788616 INVITE User-Agent: VoIPMS/SERAST Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:1MYPHONE@74.63.41.218> Content-Length: 0 ------------------------------------------------------------------------ send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500: ------------------------------------------------------------------------ ACK sip:1MYPHONE@newyork.voip.ms SIP/2.0 Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN Max-Forwards: 69 From: "Extension 1001" <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S To: <sip:1MYPHONE@newyork.voip.ms>;tag=as632cb7d9 Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a CSeq: 7788616 ACK Content-Length: 0 ------------------------------------------------------------------------ 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed. Cause: NO_ANSWER 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1001(a)192.168.1.100 has executed the last dialplan instruction, hanging up. 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING] 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/external/1MYPHONE) Ended 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/1MYPHONE [CS_DESTROY] send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750: ------------------------------------------------------------------------ SIP/2.0 503 Service Unavailable Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180 From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100>;tag=g0Qyy0ZQ96gmg Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244504 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1001(a)192.168.1.100) Ended Content-Length: 02011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1001(a)192.168.1.100 [CS_DESTROY] Remote-Party-ID: "MYPHONE" <sip:MYPHONE@192.168.1.100>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125: ------------------------------------------------------------------------ ACK sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj To: <sip:MYPHONE@192.168.1.100>;tag=g0Qyy0ZQ96gmg Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn CSeq: 5244504 ACK Supported: sec-agree Max-Forwards: 70 Proxy-Authorization: Digest qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE@192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913" Content-Length: 0 ------------------------------------------------------------------------ Thank you in advance.

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CDR Fields
by Tim King 31 Jan '11

31 Jan '11

Guys I have been digging for a while and I am struggling to find a page the lists all of the possible fields (preferably with brief descriptions) that are available with the various CDR records aka(XML and CSv). I am also curious if anyone as found a way to record quality information aka(Jitter, Latency, etc..) in the CDR record for the call. Thanks Tim

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