Shimul Kanti Barua
2003-Sep-17 03:21 UTC
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Caller ID Problems (WipeOut .) > 2. Re: IAX, IAX2 and authenticatyion (Dan) > 3. RE: 7206 as SIP->PSTN Gateway? (Abdul Hakeem) > 4. Re: IAX, IAX2 and authenticatyion (Brancaleoni Matteo) > 5. Re: Dect Phone (Tjardick van der Kraan) > 6. Monitoring an active channel (Timothy Soos) > 7. Re: asterisk and defunct perl procs (Rich Adamson) > 8. Re: Caller ID Problems (Rich Adamson) > 9. UK Suppliers (Angel Gabriel) > 10. RE: UK Suppliers (Lee Redmayne) > 11. How to test * ? (Angel Gabriel) > 12. Re: IAX, IAX2 and authenticatyion (dtoma@fx.ro) > 13. Re: UK Suppliers (YO Internet Information) > 14. Re: asterisk and defunct perl procs (Angel Gabriel) > 15. Re: asterisk and defunct perl procs (Rich Adamson) > 16. Re: Asterisk using a h323 gateway (Michael Manousos) > > --__--__-- > > Message: 1 > From: "WipeOut ." <wipeout@linuxmail.org> > To: asterisk-users@lists.digium.com > Date: Sat, 13 Sep 2003 06:41:43 +0000 > Subject: Re: [Asterisk-Users] Caller ID Problems > Reply-To: asterisk-users@lists.digium.com > > There are two things I can think of.. > > 1. You are not paying for CallerID support from your telco on that line..Its is not always a standard feature..> > 2. The CallerID that your telco provides is not compatible with the digiumcard and Asterisk..> > > > > Dear Asterisk User, > > > > I am trying to use a Digium FXO Card to get the callerid but fail. > > > > Asterisk version: Asterisk CVS-09/03/03-11:15:03 > > > > In my zapata.conf > > usecallerid=yes > > hidecallerid=no > > callwaitingcallerid=yes > > rxgain=3.0 > > txgain=3.0 > > ;callprogress=yes > > > > When I use my mobile (my mobile will show callerid) dial a call to thesystem Zap/1-1 channel. Then I use "show channel zap/1-1" The callerid field show "Caller ID: (N/A)"> > > > Please help ... Anywhere I can check and anywhere I done wrong? > > > > Thanks, > Randal > -- > ______________________________________________ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > > --__--__-- > > Message: 2 > From: "Dan" <dtoma@fx.ro> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > Date: Sat, 13 Sep 2003 09:49:13 +0300 > Organization: Personal Use > Reply-To: asterisk-users@lists.digium.com > > Hi Martin, > > ----- Original Message ----- > From: "Martin Pycko" <martinp@digium.com> > To: "Asterisk Users" <asterisk-users@lists.digium.com> > Sent: Friday, September 12, 2003 11:11 PM > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > > > > IAX2 uses 4569 UDP port. > > How this port can be changed? There is no iax2.conf file... > > Dan > > > --__--__-- > > Message: 3 > From: "Abdul Hakeem" <alhakeem@blueyonder.co.uk> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > Date: Sat, 13 Sep 2003 08:21:40 +0100 > Reply-To: asterisk-users@lists.digium.com > > Hi, > You need the PA-VFC-2TE1+ cards. It supports 60 calls for codecs such as > G723 and 120 calls for G729a and b(with the addition of a PA-MCX card). > > Cheers, > Abdul > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Kane > Sent: 12 September 2003 18:30 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > > > Also, don't limit yourself to Cisco. There are many vendors out there > that make SIP trunking gateways... > > > ----- Original Message ----- > From: "David C. Troy" <dave@toad.net> > To: <asterisk-users@lists.digium.com> > Sent: Friday, September 12, 2003 1:24 PM > Subject: [Asterisk-Users] 7206 as SIP->PSTN Gateway? > > > > > > All, > > > > I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. > > Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know > > > which cards, if any, exist for a 7206VXR to act in a similar capacity, > > > either as a T1/PRI, DS3, or POTS FXO/FXS? > > > > What other Cisco routers can act as SIP gateways today? > > > > Thanks, > > Dave > > > > ====================================================================> > David C. Troy [dave@toad.net] 410-384-2500 Sales > > ToadNet - Want to go fast? 410-544-1329 FAX > > 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > From: Brancaleoni Matteo <mbrancaleoni@espia.it> > To: asterisk-users@lists.digium.com > Organization: Espia - Emmegi Srl > Date: Sat, 13 Sep 2003 09:52:34 +0200 > Reply-To: asterisk-users@lists.digium.com > > hi. > actualy the iax2 conf file is the same of iax . > iax2 port is hardcoded in channels/iax2.h, line 72 (more or less) > You can change it & recompile. > > matteo. > > Il sab, 2003-09-13 alle 08:49, Dan ha scritto: > > Hi Martin, > > > > ----- Original Message ----- > > From: "Martin Pycko" <martinp@digium.com> > > To: "Asterisk Users" <asterisk-users@lists.digium.com> > > Sent: Friday, September 12, 2003 11:11 PM > > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > > > > > > > IAX2 uses 4569 UDP port. > > > > How this port can be changed? There is no iax2.conf file... > > > > Dan > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Brancaleoni Matteo <mbrancaleoni@espia.it> > Espia - Emmegi Srl > > > --__--__-- > > Message: 5 > From: "Tjardick van der Kraan" <tjardick@vanderkraan.net> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] Dect Phone > Date: Sat, 13 Sep 2003 10:47:29 +0200 > Reply-To: asterisk-users@lists.digium.com > > > ----- Original Message ----- > From: "Robert Boardman" <robb@boardman.me.uk> > To: <asterisk-users@lists.digium.com> > Sent: Friday, September 12, 2003 10:26 PM > Subject: [Asterisk-Users] Dect Phone > > > > Hi > > > > I have a problem with a new DECT phone I have bought > > > > The key pad works like a Mobile phone where you dial first then pick up > > the line, but it seems to dail too fast or spuriously, ie 012826736464 > > show on thew Asterisk console as 0012282677, could any one offer advice > > how to fix? > > Have you tried hitting dial before typing the numbers ? My dect does giveme> the dialtone then. > (allthough i don't have the problem that digits go to quick, but maybe you > can tweak that in an advanced menu setting on the phone). > > Tj > > > --__--__-- > > Message: 6 > From: Timothy Soos <XQL@americanisp.net> > Organization: XQL, LLC > To: asterisk-users@lists.digium.com > Date: Sat, 13 Sep 2003 05:13:54 -0600 > Subject: [Asterisk-Users] Monitoring an active channel > Reply-To: asterisk-users@lists.digium.com > > Hello All, > > I am still having some difficulty working to monitor an already active > channel. I did some experimenting with the Monitor application without > achieving my desired results. > > Here are the relevant parts of my extensions.conf file: > [CustomerSide] > exten => 2,1,StopMonitor > exten => 3,1,Monitor(wav,Test_Recording_1) > > This is what happens: > 1. From the console, I dial to the phone connected to the TDM400P card: > *CLI> dial 1234@CustomerSide > and answer the phone when it rings. > 2. Next, I dial from the console to activate monitoring: > *CLI> dial 3@CustomerSide > Unfortunately, the monitoring does not start, and I hear Asterisk sending3> DTMF tones to the phone. > > What am I doing wrong that prevents the monitoring from starting? > > Is it required to start the monitoring from another phone (hard or soft > phone)? > -- > Thanks, > Tim > > --__--__-- > > Message: 7 > Date: Sat, 13 Sep 2003 06:42:03 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] asterisk and defunct perl procs > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > FWIW, I just immplemented * on a RH9 box using the CVS without anyproblems> whatsoever. The RH9 box was built from CD's as a workstation (witheverything> installed), up2date ran to bring it reasonably current, etc. I hadinstalled> "ser" a few weeks ago and it worked properly as well. Ser was shutdown(still> remains installed) and * is running now. > > I did not have to export anything or do anything special with the systemother> then to ensure the running kernel and its "matching" source code wasinstalled.> That was required due to the Digium X100P card installation needs,otherwise> * installed and ran correctly the first time. > > ------------------------ > > Yes, this is RH9. Thank you for the info. > > > > On Fri, Sep 12, 2003 at 02:59:46PM -0700, Scott Stingel wrote: > > > If you're running RedHat 9, there is a known problem. > > > > > > Try executing the following line in the shell before startingasterisk:> > > > > > export LD_ASSUME_KERNEL=2.4.1 > > > > > > Hope this works! > > > > > > -Scott > > > > > > Scott M. Stingel > > <snip> > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ---------------End of Original Message----------------- > > > > --__--__-- > > Message: 8 > Date: Sat, 13 Sep 2003 06:56:51 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] Caller ID Problems > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > I'm having some of the same issues and it seems to be related totransmission> levels. CallerID worked fine prior to me messing with rxgain/txgain, butI've> not gone back to verify what I did to muck it up as yet. > > ------------------------ > > There are two things I can think of.. > > > > 1. You are not paying for CallerID support from your telco on thatline.. Its is not always a> standard feature.. > > > > 2. The CallerID that your telco provides is not compatible with thedigium card and Asterisk..> > > > > > > > > Dear Asterisk User, > > > > > > I am trying to use a Digium FXO Card to get the callerid but fail. > > > > > > Asterisk version: Asterisk CVS-09/03/03-11:15:03 > > > > > > In my zapata.conf > > > usecallerid=yes > > > hidecallerid=no > > > callwaitingcallerid=yes > > > rxgain=3.0 > > > txgain=3.0 > > > ;callprogress=yes > > > > > > When I use my mobile (my mobile will show callerid) dial a call to thesystem Zap/1-1 channel.> Then I use "show channel zap/1-1" The callerid field show "Caller ID:(N/A)"> > > > > > Please help ... Anywhere I can check and anywhere I done wrong? > > > > --__--__-- > > Message: 9 > From: Angel Gabriel <badmangabriel@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:01:32 +0100 > Subject: [Asterisk-Users] UK Suppliers > Reply-To: asterisk-users@lists.digium.com > > Can anyone please direct me to UK based suppliers of equipment. Website > URL's would be appreciated. TIA > -- > ***** > Not everyone is touched by an Angel.... > .... Those that are, never forget the experience > ***** > > > --__--__-- > > Message: 10 > From: "Lee Redmayne" <lee.redmayne@nwva.org> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] UK Suppliers > Date: Sat, 13 Sep 2003 13:11:52 +0100 > Reply-To: asterisk-users@lists.digium.com > > I bought some Snom phones which work nicely with Asterisk from: > > ProVu Communications Ltd > Bank House > Marsden > Huddersfield > HD7 6BR > > 01484-840048 > info@provu.co.uk > www.provu.co.uk > > -----Original Message----- > From: Angel Gabriel > Sent: 13 September 2003 13:02 > To: * Users > Subject: [Asterisk-Users] UK Suppliers > > Can anyone please direct me to UK based suppliers of equipment. Website > URL's would be appreciated. TIA > -- > ***** > Not everyone is touched by an Angel.... > .... Those that are, never forget the experience > ***** > > > --__--__-- > > Message: 11 > From: Angel Gabriel <badmangabriel@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:22:11 +0100 > Subject: [Asterisk-Users] How to test * ? > Reply-To: asterisk-users@lists.digium.com > > I was wondering, can I test * using just a modem card? I was want to > check ome of the features, before I go and buy some cards. (Thanks for > th elink to the reseller page, you know who you are!) > -- > ***** > Not everyone is touched by an Angel.... > .... Those that are, never forget the experience > ***** > > > --__--__-- > > Message: 12 > Date: Sat, 13 Sep 2003 15:27:31 +0300 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion > From: dtoma@fx.ro > Reply-To: asterisk-users@lists.digium.com > > At Sat, 13 Sep 2003 09:52:34 +0200 , asterisk-users@lists.digium.comwrote:> > >hi. > >actualy the iax2 conf file is the same of iax . > >iax2 port is hardcoded in channels/iax2.h, line 72 (more or less) > >You can change it & recompile. > > > >matteo. > > > > Thanks a lot. > Dan > ... > > --__--__-- > > Message: 13 > From: "YO Internet Information" <tan@yointernet.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] UK Suppliers > Date: Sat, 13 Sep 2003 13:30:41 +0100 > Organization: YO Internet Services Ltd > Reply-To: asterisk-users@lists.digium.com > > http://www.telappliant.co.uk > > > > > ----- Original Message ----- > From: "Lee Redmayne" <lee.redmayne@nwva.org> > To: <asterisk-users@lists.digium.com> > Sent: Saturday, September 13, 2003 1:11 PM > Subject: RE: [Asterisk-Users] UK Suppliers > > > I bought some Snom phones which work nicely with Asterisk from: > > ProVu Communications Ltd > Bank House > Marsden > Huddersfield > HD7 6BR > > 01484-840048 > info@provu.co.uk > www.provu.co.uk > > -----Original Message----- > From: Angel Gabriel > Sent: 13 September 2003 13:02 > To: * Users > Subject: [Asterisk-Users] UK Suppliers > > Can anyone please direct me to UK based suppliers of equipment. Website > URL's would be appreciated. TIA > -- > ***** > Not everyone is touched by an Angel.... > .... Those that are, never forget the experience > ***** > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 14 > Subject: Re: [Asterisk-Users] asterisk and defunct perl procs > From: Angel Gabriel <badmangabriel@lycos.co.uk> > To: * Users <asterisk-users@lists.digium.com> > Date: 13 Sep 2003 13:24:52 +0100 > Reply-To: asterisk-users@lists.digium.com > > On Sat, 2003-09-13 at 13:42, Rich Adamson wrote: > > FWIW, I just immplemented * on a RH9 box using the CVS without anyproblems> > whatsoever. The RH9 box was built from CD's as a workstation (witheverything> > installed), up2date ran to bring it reasonably current, etc. I hadinstalled> > "ser" a few weeks ago and it worked properly as well. Ser was shutdown(still> > remains installed) and * is running now. > > At the risk of sounding dumb, what's ser ? > -- > ***** > Not everyone is touched by an Angel.... > .... Those that are, never forget the experience > ***** > > > --__--__-- > > Message: 15 > Date: Sat, 13 Sep 2003 07:49:22 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] asterisk and defunct perl procs > To: asterisk-users@lists.digium.com > Reply-To: asterisk-users@lists.digium.com > > > > > FWIW, I just immplemented * on a RH9 box using the CVS without anyproblems> > > whatsoever. The RH9 box was built from CD's as a workstation (witheverything> > > installed), up2date ran to bring it reasonably current, etc. I hadinstalled> > > "ser" a few weeks ago and it worked properly as well. Ser was shutdown(still> > > remains installed) and * is running now. > > > > At the risk of sounding dumb, what's ser ? > > From the 20,000 foot level: > > Asterisk is a PBX with lots of local features > > Ser is the Central Office switch ( http://www.iptel.org/ser/ ) > > If you had hundreds/thousands of users and/or pbx's, ser typically handles > the call routing. FWD uses ser as an example. Both are mostly opensource.> > > > > --__--__-- > > Message: 16 > Date: Sat, 13 Sep 2003 16:32:32 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Organization: inAccess Networks > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway > Reply-To: asterisk-users@lists.digium.com > > Cerrajetto wrote: > > Hello: > > > > I am testing Asterisk with oh323. > > > > My question is: can Asterisk route some calls thru a second h323 gateway(a> > h323 <-> PSTN gw)? > > > > - Asterisk ip: 192.168.1.10 > > - h323<->PSTN gw: 192.168.1.20 > > > > I've tried: > > > > exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) > > > > or > > > > exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) > > I guess that "192.1.1.20" is a typo, right? > You will have to give more info in order to be able to > find the problem. > Try to set these params in oh323.conf file: > > wrapLibTraceLevel=3 > libTraceLevel=3 > libTraceFile=/tmp/trace.txt > > Rerun and send me the "/tmp/trace.txt" file, "oh323.conf" > and the screen log (off-list). > > > > > but it does not work at all. > > > > If my h323 client directly uses 192.168.1.20 as h323 gateway, the callsare> > routed to the PSTN perfectly. > > > > What is the correct way to route some calls from Asterisk to anotherh323> > gateway? > > > > Thank you, > > Mark > > > > > Michael. >Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream ------------ exten => _01XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) ; Crilium --------- exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) Shimul> > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >
Josh Roberson
2003-Sep-17 04:12 UTC
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
This may just be me, but When replying to a message from a digest, it would be proper to remove all the context except that to which you are replying so as not to have to scroll an entire mile to see your reply. I know if I was the person you were replying to, I probably wouldn't scroll all the way through the other 15 messages just to see a reply. Just my .02, Sorry if I seem a bit irrational, just irritated. -Josh -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Shimul Kanti Barua Sent: Wednesday, September 17, 2003 4:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs <<<<<BIG OLE SNIP>>>>>>>> Message: 16 > Date: Sat, 13 Sep 2003 16:32:32 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Organization: inAccess Networks > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway > Reply-To: asterisk-users@lists.digium.com > > Cerrajetto wrote: > > Hello: > > > > I am testing Asterisk with oh323. > > > > My question is: can Asterisk route some calls thru a second h323gateway (a> > h323 <-> PSTN gw)? > > > > - Asterisk ip: 192.168.1.10 > > - h323<->PSTN gw: 192.168.1.20 > > > > I've tried: > > > > exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) > > > > or > > > > exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) > > I guess that "192.1.1.20" is a typo, right? > You will have to give more info in order to be able to > find the problem. > Try to set these params in oh323.conf file: > > wrapLibTraceLevel=3 > libTraceLevel=3 > libTraceFile=/tmp/trace.txt > > Rerun and send me the "/tmp/trace.txt" file, "oh323.conf" > and the screen log (off-list). > > > > > but it does not work at all. > > > > If my h323 client directly uses 192.168.1.20 as h323 gateway, thecalls are> > routed to the PSTN perfectly. > > > > What is the correct way to route some calls from Asterisk to anotherh323> > gateway? > > > > Thank you, > > Mark > > > > > Michael. >Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream ------------ exten => _01XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) ; Crilium --------- exten => _9XXXXXXXXXX,1,Dial(OH323/BYEXTENSION@xxx.xxx.xxx.xxx) Shimul> > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003