April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

IVR
by azza miled 10 Apr '12

10 Apr '12

Hi I want to make an IVR with lua. I am e real begginer and I don't know where to place my script lua ( the path) to be functionnel. Can any one help please (any intersting tutoriel can help) Thanks

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FreeSWITCH Conference Call Tomorrow, April 11
by Michael Collins 10 Apr '12

10 Apr '12

Hello all! Tomorrow's agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_04_11 We are going to be talking about some "what if" scenarios. Feel free to bring your own, but the two specifically that we'll be discussing tomorrow: * What if I host my FS-related project on GitHub? * What if I wanted to build a desk phone with cheap ARM hardware? Please join the conversation and bring your expertise as well as your questions for discussion. Talk to you tomorrow! -Michael

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gsmopen and USB dongle
by Giovanni Maruzzelli 10 Apr '12

10 Apr '12

Hi FreeSWITCHers, I'm pleased to tell you I'm about to release an updated version of mod-gsmopen (cellular network voice endpoint and trunking, and sms gateway). Most important feature will be support for the cheap and very reliable huawei family of usb dongles, eg: Huawei E1550 (no need for soundcard, no cpu load, native pcm stream). So, get ready, and start looking on ebay or your retail shop for those E1550 dongles (roughly $20 one item), and start testing the updated gsmopen module as soon as is released. Dozen of concurrent gsm channels (calls) support ! Stay tuned for further announcements in some days. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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mod_spandsp
by Malay Thakershi 10 Apr '12

10 Apr '12

Is mod_spandsp only for fax? Does it require special hardware? Can I use codec it provides for regular calls? Thanks. Malay

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ClueCon 2012: Call For Speakers
by Michael Collins 10 Apr '12

10 Apr '12

[image: Inline image 2] ClueCon 2012 - Call For Speakers ClueCon - the open source telephony conference by developers, for developers - would like to announce that we are having an open call for speaking proposals for this year's event. If you have an idea for a technical presentation for ClueCon 2012 then we would like to hear about. What makes a great ClueCon presentation? The tech savvy crowd that attends ClueCon loves *technical* presentations. In general, the more technical the presentation, the better. If you are thinking about a presentation then consider these points: - ClueCon talks are 30 minutes in length, including Q&A time with the audience - ClueCon has a special focus on open source VoIP and telephony projects like FreeSWITCH, Asterisk, OpenSIPS, and Kamailio - Attendees enjoy hearing about projects built with open source tools - Highly technical discussions that show the nuts and bolts are especially well-liked - The audience appreciates seeing and participating in live demonstrations Please send your proposals to marketing(a)cluecon.com. Be sure to include a working title, description of the talk, and name of the presenter. Don't delay! There are a limited number of openings. ClueCon 2012 Registration Information ClueCon 2012 registration is now open. Visit the registration page<http://www.cluecon.com/register?cc12cfs>for details. Be sure to book your room at the Wyndham <http://www.cluecon.com/hotels?cc12cfs> and qualify for the $300 discount. As always, feel free to call us at 877.742.CLUE (877.742.2583) if you have any questions about ClueCon 2012. Also, keep in mind that the FreeSWITCH community has a conference call<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12cfs>each Wednesday at 1PM Eastern time. This is a great opportunity to talk about open source telephony and get to know a number folks who will be at ClueCon 2012. Stay tuned for more news about ClueCon speakers, sponsors, and related events! -- ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12cfs1

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Strange DTMF Tones On Inbound Calls
by Daniel Knaggs 10 Apr '12

10 Apr '12

Hello all, Got a bit of a strange one, we appear to be getting DTMF tones on incoming calls when the caller hasn’t even pressed any keys. It normally happens with 10 seconds or so after the call has been answered. Here is the log of it happening earlier: - 2012-03-23 10:38:16.852654 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF A (debug = 0) 2012-03-23 10:38:16.852654 [DEBUG] mod_freetdm.c:799 Queuing DTMF [A] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:16.915653 [DEBUG] switch_rtp.c:2420 Send start packet for [A] ts=49440 dur=160/160/2000 seq=46803 lw=49440 2012-03-23 10:38:16.936653 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=320/320/2000 seq=46804 lw=49600 2012-03-23 10:38:16.957652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=480/480/2000 seq=46805 lw=49760 2012-03-23 10:38:16.978652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=640/640/2000 seq=46806 lw=49920 2012-03-23 10:38:16.999652 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=800/800/2000 seq=46807 lw=50080 2012-03-23 10:38:17.020651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=960/960/2000 seq=46808 lw=50240 2012-03-23 10:38:17.041651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1120/1120/2000 seq=46809 lw=50400 2012-03-23 10:38:17.062651 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1280/1280/2000 seq=46810 lw=50560 2012-03-23 10:38:17.083650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1440/1440/2000 seq=46811 lw=50720 2012-03-23 10:38:17.104650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1600/1600/2000 seq=46812 lw=50880 2012-03-23 10:38:17.125650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1760/1760/2000 seq=46813 lw=51040 2012-03-23 10:38:17.146650 [DEBUG] switch_rtp.c:2323 Send middle packet for [A] ts=49440 dur=1920/1920/2000 seq=46814 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46815 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46816 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2323 Send end packet for [A] ts=49440 dur=2080/2080/2000 seq=46817 lw=51200 2012-03-23 10:38:17.167661 [DEBUG] switch_rtp.c:2271 Queue digit delay of 40ms 2012-03-23 10:38:18.070638 [DEBUG] ftdm_io.c:3530 [s1c2][1:2] Queuing DTMF B (debug = 0) 2012-03-23 10:38:18.070638 [DEBUG] mod_freetdm.c:799 Queuing DTMF [B] in channel FreeTDM/1:2/000 device 1:2 2012-03-23 10:38:18.133637 [DEBUG] switch_rtp.c:2420 Send start packet for [B] ts=59040 dur=160/160/2000 seq=46864 lw=59040 2012-03-23 10:38:18.154637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=320/320/2000 seq=46865 lw=59200 2012-03-23 10:38:18.175637 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=480/480/2000 seq=46866 lw=59360 2012-03-23 10:38:18.196636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=640/640/2000 seq=46867 lw=59520 2012-03-23 10:38:18.217636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=800/800/2000 seq=46868 lw=59680 2012-03-23 10:38:18.238636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=960/960/2000 seq=46869 lw=59840 2012-03-23 10:38:18.259636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1120/1120/2000 seq=46870 lw=60000 2012-03-23 10:38:18.280636 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1280/1280/2000 seq=46871 lw=60160 2012-03-23 10:38:18.301635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1440/1440/2000 seq=46872 lw=60320 2012-03-23 10:38:18.322635 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1600/1600/2000 seq=46873 lw=60480 2012-03-23 10:38:18.343634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1760/1760/2000 seq=46874 lw=60640 2012-03-23 10:38:18.364634 [DEBUG] switch_rtp.c:2323 Send middle packet for [B] ts=59040 dur=1920/1920/2000 seq=46875 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46876 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46877 lw=60800 2012-03-23 10:38:18.385634 [DEBUG] switch_rtp.c:2323 Send end packet for [B] ts=59040 dur=2080/2080/2000 seq=46878 lw=60800 I’m very sure the caller isn’t dialling “A” or “B”! Running on ISDN via a TE121 card using E1 (euroisdn). Freeswitch version is “FreeSWITCH Version 1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)”. Currently there is a “bind_meta_app” in the config which binds a script on the B leg of the call which parks the call – I haven’t tried turning this off yet to see if it’s this. Wondering if anyone has any ideas or has come across this before? Thanks in advance. [cid:imageacd695.PNG@9e92e461.40aa4b96] Daniel Knaggs Software Developer Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE Tel: 01482 828000 / Fax: 01482 373100 Daniel.Knaggs(a)realitysolutions.co.uk<mailto:Daniel.Knaggs@realitysolutions.co.uk> www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/> ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image27a71e.PNG@c2da8488.4683bff1]

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Skypopen failurereason 7 vaa_input_status false
by Marcin Gozdalik 10 Apr '12

10 Apr '12

Hi I am trying to get mod_skypopen to work with my FS installation. I used interactive installer to setup skypopen. It mostly works OK, but the download path is dead now, and you have to get Skype 2.0.0.72 binary from elsewhere. When starting up Skype clients I get: ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi ALSA lib ../../../src/pcm/pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi I got it up to a point where I have 2 Skypes on one login running. mod_skypopen loads and connects to those instances with no problem. mod_skypopen loads correctly and I can see them as connected: freeswitch@internal> sk list sk console is NOT yet assigned F ID Name IB (F/T) OB (F/T) State CallFlw UUID = ==== ======== ======= ======= ====== ============ ====== 1 [skype101] 0/0 1/1 IDLE IDLE * 2 [skype102] 0/0 0/0 IDLE IDLE Total Interfaces: 2 IB Calls(Failed/Total): 0/0 OB Calls(Failed/Total): 1/1 I can even send Skype chat messages. When I try to make a voice call I get: 2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:173 [406c21c|1b4c78b] [DEBUG_SKYPE 173 ][skype101 ][SELECTD,IDLE] READING: |||CALL 33 STATUS UNPLACED||| 2012-04-10 13:41:16.998767 [DEBUG] skypopen_protocol.c:718 [406c21c|1b4c78b] [DEBUG_SKYPE 718 ][skype101 ][DIALING,UNPLACD] skype_call: 33 is now UNPLACED 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173 [406c21c|1b4c78b] [DEBUG_SKYPE 173 ][skype101 ][DIALING,UNPLACD] READING: |||CALL 33 STATUS ROUTING||| 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:713 [406c21c|1b4c78b] [DEBUG_SKYPE 713 ][skype101 ][DIALING,ROUTING] skype_call: 33 is now ROUTING 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173 [406c21c|1b4c78b] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 33 FAILUREREASON 7||| 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:540 [406c21c|1b4c78b] [DEBUG_SKYPE 540 ][skype101 ][DIALING,ROUTING] Skype FAILED on skype_call 33. Let's wait for the FAILED message. 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173 [406c21c|1b4c78b] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 33 VAA_INPUT_STATUS FALSE||| 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:173 [406c21c|1b4c78b] [DEBUG_SKYPE 173 ][skype101 ][DIALING,ROUTING] READING: |||CALL 33 STATUS FAILED||| 2012-04-10 13:41:17.088638 [DEBUG] skypopen_protocol.c:687 [406c21c|1b4c78b] [DEBUG_SKYPE 687 ][skype101 ][DIALING,FAILED] we tried to call Skype on skype_call 33 and Skype has now FAILED 2012-04-10 13:41:17.088638 [DEBUG] mod_skypopen.c:1420 [406c21c|1b4c78b] [DEBUG_SKYPE 1420 ][skype101 ][DOWN,FAILED] skype call ended When trying to make a call from "normal" Skype client to skypopen I received message, that the other end uses too old a Skype client and I can't call him. So I upgraded the Skype binary to 2.2.0.35 but no luck either. Skype processes run as root, and I have the skypopen module loaded and /dev/dsp present: root@conf1:~# lsmod | grep skyp skypopen 4735 0 root@conf1:~# ls -al /dev/dsp crw-r--r-- 1 root root 14, 3 2012-04-06 17:13 /dev/dsp I am running 64 bit Ubuntu 10.04.1 (all latest security applied). FS is FreeSWITCH Version 1.0.head (git-bda1a8d 2012-03-13 15-42-29 +0100) What am I missing? As a sidenote, does Skype 2.2 support OSS sound at all? If not than mod_skypopen became one-way only as Skype refuses to make a call to Skype 2.0 Linux client. -- Marcin Gozdalik

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mod_nibblebill issue - channel is not initialized for billing
by Ali Jawad 10 Apr '12

10 Apr '12

Hi I have a kamailio server that sends calls to FS, calls then get routed to a VOIP provider. This works just fine, but calls dont get billed and it seems billing is not initialized for the calls at hand Below is the output for the FS console for such a call : freeswitch@fs-100-151> 2012-04-10 13:52:16.102555 [NOTICE] switch_channel.c:926 New Channel sofia/internal/support1(a)Kamailio.IP [e41ebdb3-eac2-4104-bad2-ed802bb8e26f] 2012-04-10 13:52:16.102555 [INFO] mod_dialplan_xml.c:485 Processing support1 <support1>->0091111111 in context public 2012-04-10 13:52:16.102555 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/support1(a)kamailio.ip to XML[009611111111@default] 2012-04-10 13:52:16.102555 [INFO] mod_dialplan_xml.c:485 Processing support1 <support1>->009611111111 in context default 2012-04-10 13:52:16.122486 [NOTICE] switch_channel.c:926 New Channel sofia/external/85719611111111 [92118bd1-cddd-4854-b303-2525212961f6] 2012-04-10 13:52:17.503477 [NOTICE] sofia_glue.c:4040 Pre-Answer sofia/external/85719611111111! 2012-04-10 13:52:17.503477 [INFO] switch_ivr_originate.c:3215 Sending early media 2012-04-10 13:52:17.503477 [NOTICE] mod_sofia.c:2585 Pre-Answer sofia/internal/support1(a)kamailio.ip! 2012-04-10 13:52:17.863305 [INFO] switch_rtp.c:3207 Auto Changing port from ip1:56854 to ip2:56854 2012-04-10 13:52:20.143135 [NOTICE] sofia.c:6214 Channel [sofia/external/12349611111111] has been answered 2012-04-10 13:52:20.163119 [NOTICE] switch_ivr_bridge.c:417 Channel [sofia/internal/support1(a)Kamailio.IP] has been answered I did check the active calls and checked billing status it says not initialized for call, see : freeswitch@fs-100-151> nibblebill e41ebdb3-eac2-4104-bad2-ed802bb8e26f check 2012-04-10 13:53:44.602951 [INFO] mod_nibblebill.c:759 Can't check - channel is not initialized for billing! Unloading and loading mod_nibblebill occurs without any errors, see : freeswitch@fs-100-151> unload mod_nibblebill +OK 2012-04-10 13:57:05.043193 [NOTICE] switch_loadable_module.c:910 Deleting Application 'nibblebill' 2012-04-10 13:57:05.043193 [NOTICE] switch_loadable_module.c:963 Deleting API Function 'nibblebill' 2012-04-10 13:57:05.043193 [CONSOLE] switch_loadable_module.c:1784 Stopping: mod_nibblebill 2012-04-10 13:57:05.043193 [NOTICE] switch_event.c:1935 Event Binding deleted for mod_nibblebill:SESSION_HEARTBEAT 2012-04-10 13:57:05.043193 [CONSOLE] switch_loadable_module.c:1804 mod_nibblebill unloaded. freeswitch@fs-100-151> load mod_nibblebill 2012-04-10 13:57:09.462673 [INFO] mod_nibblebill.c:228 Connected ODBC DSN: bandwidth.com 2012-04-10 13:57:09.462673 [CONSOLE] switch_loadable_module.c:1318 Successfully Loaded [mod_nibblebill] 2012-04-10 13:57:09.462673 [NOTICE] switch_loadable_module.c:254 Adding Application 'nibblebill' 2012-04-10 13:57:09.462673 [NOTICE] switch_loadable_module.c:298 Adding API Function 'nibblebill' +OK Reloading XML +OK freeswitch@fs-100-151> 2012-04-10 13:57:09.462673 [INFO] mod_enum.c:812 ENUM Reloaded 2012-04-10 13:57:09.462673 [INFO] switch_time.c:1128 Timezone reloaded 530 definitions My dialplan includes the following : <include> <extension name="Dial Out to Custom GW"> <condition field="destination_number" expression="(\d{10})$"> <action application="set" data="nibble_account=${accountcode}"/> <action application="set" data="nibble_rate=0.30"/> <action application="bridge" data="sofia/gateway/custom/1234$1"/> </condition> </extension> </include> Thanks

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skypopen mod wont let me call out
by millerman33 10 Apr '12

10 Apr '12

Every time I place a call in skype through freeswitch I keep getting this error message freeswitch@Home-PC>2012-04-07 13:11.26.261968 [ERR] mod_skyopen.c:2330 [git-] [ERRORA 2330 ][interface1 ][RINGING,RINGING] No session_uuid_str??? Thats every outbound call, im a newbie to this maybe I installed something wrong. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skypopen-mod-wont-let-me-call… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Cookbook example
by Miha 10 Apr '12

10 Apr '12

Hi, I need a little help. I am trying to use example for inserting XML to mysql cdrtable. I have add id before caller_id_name as I need Id in mysql table. Problem is that I am getting random numbers in ID field, sometimes values for ID are the same, so sql perl script return error and values are not inserted. Can you help me how can I modify so that this will work properly. use strict; use warnings; use CGI; use DBI; use Data::Dump qw(dump); $|++; my $q = CGI->new; my $raw_cdr = $q->param('cdr'); my @all_fields = qw(id caller_id_name caller_id_number destination_number context start_stamp answer_stamp end_stamp duration billsec hangup_cause uuid bleg_uuid accountcode domain_name); my @fields; my @values; foreach my $field (@all_fields) { next unless $raw_cdr =~ m/$field>(.*?)</; push @fields, $field; push @values, "'" . urldecode($1) . "'"; } my $cdr_line; my $query = sprintf( "INSERT INTO %s (%s) VALUES (%s);", 'radacct_1', join(',', @fields), join(',', @values) ); my $db = DBI->connect('DBI:mysql:dbname=voipbill_slo;host=xxx.xxx.xxx.xxx','xxxxxx', 'xxxxx'); $db->do($query); print $q->header(); sub urldecode { my $url = shift; $url =~ s/%([a-fA-F0-9]{2,2})/chr(hex($1))/eg; return $url; } [root@fs1 cgi-bin]# cat /var/log/secure Thank you! Regards, Miha

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