Re: [Freeswitch-users] Resend: Issues with SIP + TCP?

Woof! On Mon, 26 Oct 2009 16:03:59 -0400, Michael Collins m...@freeswitch.org wrote: I was wondering... does anyone make a SIP certification program kinda like a pen-tester except to find all the ways your SIP setup is broken? Just curious. Here is a start: http://interop.sipxecs.org/

Re: [Freeswitch-users] check to see if freeswitch is alive

Woof! On Mon, 19 Oct 2009 12:07:10 -0400, Christian Löschenkohl christian.loeschenk...@xpirio.com wrote: any ideas would be helpful We run a perl script that checks if the servers are responding to requests. It can send OPTIONS, and PING requests to various servers periodically. If the

Re: [Freeswitch-users] NAT problems - sorry

be added to the WIKI? Cheers Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 07 October 2009 19:07 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] NAT problems

[Freeswitch-users] NAT problems - sorry

thanks Andy ___ FreeSWITCH-users mailing list 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

Re: [Freeswitch-users] NAT problems - sorry

BUT.. nat_map status, gives: API CALL [nat_map(status)] output: false And there is no mention of nat detection in the startup log. Is this because I'm using port 5060 externally? Cheers Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun

Re: [Freeswitch-users] NAT problems - sorry

a upnp/nat-pmp router so you'll have to manually forward everything... All the info you showed displaying the profile status is correct. /b On Oct 7, 2009, at 12:44 PM, Andy wrote: Is this because I'm using port 5060 externally? Cheers Andy

Re: [Freeswitch-users] NAT problems - sorry

misunderstood what is required. Is there some additional forwarding within FS required. I'm really sorry to keep coming back but I've been wrestling with this for a long time now and not getting anywhere. Many thanks Andy _ From: freeswitch-users-boun...@lists.freeswitch.org

[Freeswitch-users] copy and past oops in mod_event_socket.c

Woof! Too simple to open a JIRA with a patch (and it actually works as written): 1804: } else if (!strncasecmp(cmd, nolinger, 6)) { That should be an 8 as nolinger is 8 characters long. --Woof! ___ Freeswitch-users mailing list

Re: [Freeswitch-users] Problems with Ping and re-registering brokengateways

-Length: 0 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2810 Ping failed voiptalk.org 2009-07-14 10:07:25.483507 [WARNING] sofia.c:2813 Unregister voiptalk.org Andy _ From: freeswitch-users-boun

[Freeswitch-users] Problems with Ping and re-registering broken gateways

restart' or restart the software this fixes the problem My questions are: 1) Why would the ping fail when the registration appears to be intact? 2) Whay would the auto re-register not work but a restart would? This ones driving me nuts so any help greatly appreciated. regards Andy

[Freeswitch-users] Question on auth-calls

Woof! It is my understanding, that if I set param name=auth-calls value=false/ in a SIP profile, it shouldn't challenge for authentication under any circumstances. However, if an INVITE contains a a Proxy-Authorization header from another proxy, Sofia DOES challenge with a 407. I'm

Re: [Freeswitch-users] Question on auth-calls

Woof! On Fri, 10 Jul 2009 12:09:05 -0400, Anthony Minessale anthony.miness...@gmail.com wrote: The way it works by default is that if you send a www-authenticate, we *always* try to process it. HOWEVER, we have a accept-blind-auth sofia profile param (in fact it was invented just for sipX)

Re: [Freeswitch-users] CDR loss possibility if FS freezes?

On Fri, 26 Jun 2009 00:51:34 -0400, Raymond Chandler intralan...@freeswitch.org wrote: windmills Just set them up to be right next to the output of the system's fan! I gotta get to the patent office, quick! --Woof! ___ Freeswitch-users mailing

Re: [Freeswitch-users] Sample rate and recordFile

of the base call to 11025? Many thanks for sorting this one for me and for all your help. regards Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 June 2009 18:52 To: freeswitch-users

Re: [Freeswitch-users] Sample rate and recordFile

of concurrent calls. Many thanks Andy _ From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 June 2009 18:11 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Sample rate

Re: [Freeswitch-users] Sample rate and recordFile

but not arbitrary rates for recording otherwise. /b On Jun 12, 2009, at 9:37 AM, Andy wrote: Hi, Sorry but I just can't find this in the documentation. I'm using recordFile to record incoming messages. I'd like the audio files produced to be 11025Hz rather than 8kHz is this possible? What setting

[Freeswitch-users] Error causing freeswitch to crash

Hi, Every few days I'm getting this error which is causing Freeswitch to crash. Can anyone tell me what may be causing this or how to prevent it? 2009-05-25 10:37:38 [CRIT] switch_core_state_machine.c:263 handle_fatality() Caught signal 11 for unmapped thread! Many thanks Andy

Re: [Freeswitch-users] Freeswitch Drops Call after 30 Seconds

VOIPTALK. Cheers Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of adamF Sent: 17 May 2009 19:12 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch Drops Call after 30

[Freeswitch-users] crash-protection and monit

freeswitch install seems to crash every few days. Also, does anyone have an example of the monit setup for freeswitch to restart it when it fails? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http

Re: [Freeswitch-users] DTMF not comming through on some calls

...@194.145.190.143 entering state [completed][200] 2009-05-15 09:47:45 [DEBUG] sofia.c:2885 sofia_handle_sip_i_state() Channel sofia/external/07540526...@194.145.190.143 entering state [ready][200] These lines appear for calls that work and not when they don't. Hope that helps. Cheers Andy -Original Message

Re: [Freeswitch-users] Install without example configurations

Woof! On Fri, May 8, 2009 at 4:22 PM, Lon Baker l...@kickasspixels.com wrote: A truly clean install out of the gate. We took a different approach. Rather than change what FS comes with, we created an alternate configuration area and point FreeSWITCH to it when we start. Here's the script:

[Freeswitch-users] Using recordFile with Icecast - looses the end of the call

/myaudio.mp3) the end of the call is missing off the resultant mp3 file. A wild shot in the dark I know but does anyone have any experience of this and how it might be resolved? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch-users

Re: [Freeswitch-users] Calls being cut off while recording a message

Hi Brian, Is NAT a known problem? Is there a work around? The messages on the lists seem to imply other folks have this working ok behind NAT firewalls. What's your recommendation for how I should proceed? regards Andy -Original Message- From: freeswitch-users-boun

Re: [Freeswitch-users] Calls being cut off while recording a message

if this is obvious but what have I done wrong? Thanks for your help Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] Calls being cut off while recording a message

related. Any clues? regards Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users

Re: [Freeswitch-users] Calls being cut off while recording a message

? regards Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 06 April 2009 14:24 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls being cut off while

Re: [Freeswitch-users] Calls being cut off while recording a message

Hi Brian, The freeswitch server is connect to the internet via a Cisico ASA firewall currently running in NAT mode. I believe it's that simple but can't be sure of the equipment between my firewall and the internet. regards Andy -Original Message- From: freeswitch-users-boun

Re: [Freeswitch-users] Calls being cut off while recording a message

if this is obvious but what have I done wrong? Thanks for your help Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:40 To: freeswitch-users@lists.freeswitch.org

[Freeswitch-users] Calls being cut off while recording a message

that intermittently, calls keep getting cut off after a number of seconds. I've attached a snapshot of the log at the point that the call gets cut off, can anyone suggest why this is happening or how I can prevent it? Many thanks Andy 2009-03-30 11:14:43 [DEBUG] switch_ivr_play_say.c:272 switch_ivr_phrase_macro

Re: [Freeswitch-users] Calls being cut off while recording a message

Hi Brian, 1.03 Thanks Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 31 March 2009 14:27 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Calls

Re: [Freeswitch-users] Losing Gateway registration

is the main issue. do you have any further tips to make this more stable and prevent the call cut off? Many thanks Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Mathieu Rene Sent: 18 March

Re: [Freeswitch-users] sip cancel request fails

Woof! Appears to be a recently fixed * bug: 0014431: Bad branch parameter value in CANCEL request http://bugs.digium.com/view.php?id=14431 --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] Losing Gateway registration

. Many thanks for your help. Andy -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: 18 March 2009 14:08 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Losing

[Freeswitch-users] Losing Gateway registration

it to reconnect with a sofia restart. I'm using the same provider and user account as with the old version of the software. Can you suggest any reaosn why this may be happening and how I can prevent it? Many thanks Andy ___ Freeswitch-users mailing list Freeswitch

Re: [Freeswitch-users] About FreeSwitch

Woof! On Fri, 06 Mar 2009 01:55:43 -0500, Vikas Sharma vikas.sharma...@gmail.com wrote: Can it be integrated with other pbx as a media server? Yep, it sure can: http://sipx-wiki.calivia.com/index.php/SipXivr --Woof! ___ Freeswitch-users mailing

Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

Woof! On Sun, 01 Mar 2009 21:28:18 -0500, Brian West br...@freeswitch.org wrote: NO. You want something that people THINK exists and works well... Reliable human/voice detection doesn't exist in ANY form. I beg to differ. See http://www.freepatentsonline.com/5521967.html for one way to do

Re: [Freeswitch-users] Detecting the origin of voice activity using VAD

Woof! On Mon, 02 Mar 2009 19:32:53 -0500, Steve Underwood ste...@coppice.org wrote: I just had a look through that patent. Its amazing. There is a lot of focussed descriptive text, but a patent only really consists of its claims. Those claims are astonishingly open-ended, and characterise

Re: [Freeswitch-users] segfault when shoutcast plays mp3 and extension hangs up

Woof! On Thu, 12 Feb 2009 17:20:18 -0500, Anthony Minessale anthony.miness...@gmail.com wrote: So I wonder what about the distro you are using that makes the same exact code not work? maybe the GCC ? Possibly. A recent (last year?) GCC change caused some order of operations to change,

[Freeswitch-users] recordFile bitrate

Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http

Re: [Freeswitch-users] recordFile bitrate

8, 2009, at 8:37 AM, Andy Ayers wrote: Hi, Is the bitrate, sample rate or format of the audio stream created by session.recordFile configurable at all? Apologies if I've missed something in the docs. cheers Andy ___ Freeswitch-users mailing list

[Freeswitch-users] Global limit on the number of conference legs

Woof! Does FS have a way of limiting the total number of conference legs on a box? I am aware that each individual conference profile can have a max-members param, but what I'm looking for would span multiple conferences, with a maximum leg limit per server, regardless of the per

Re: [Freeswitch-users] Global limit on the number of conference legs

Woof! On Wed, 07 Jan 2009 18:41:31 -0500, Michael Jerris m...@jerris.com wrote: you can however use mod_limit to implement this yourself with dialplan logic as long as it is used before all calls to the conference (it wouldn't work for outbound calls from the conference without a little bit

Re: [Freeswitch-users] sofia deflect issue

Woof! On Mon, 05 Jan 2009 23:43:26 -0500, jonathan augenstine jaugenst...@gmail.com wrote: According to RFC 3515 there are no BYE messages in the protocol exchange. Once the REFER is completed (as determined by a final response returned in the NOTIFY SIPFRAG from the REFER), the original

[Freeswitch-users] DTMF and firewall

/router? The router is currently set to allow all traffic. Many thanks for any help you can give. regards Andy ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

Re: [Freeswitch-users] Where FreeSWITCH writes some files

Woof! On Sat, 13 Dec 2008 18:52:59 -0500, Michael Collins m...@freeswitch.org wrote: All you'd have to do is modify the logfile.conf.xml file and pick a new path for your freeswitch.log file... I agree. I had discovered this option and considered it as a workaround. Then I also found that

Re: [Freeswitch-users] Where FreeSWITCH writes some files

Woof! On Mon, 15 Dec 2008 11:56:53 -0500, Brian West br...@freeswitch.org wrote: I can't figure out why the log file would need to be in the db folder... I think you misunderstand. It's these files: freeswitch.history freeswitch.pid freeswitch.xml.fsxml That I feel would be better

Re: [Freeswitch-users] Where FreeSWITCH writes some files

Woof! On Mon, 15 Dec 2008 13:16:32 -0500, Raymond Chandler intralan...@freeswitch.org wrote: if freeswitch.history isn't a log, what is it? seems to me taht it's a log of what commands you've run recently... it's definitely NOT a database Actually, I the readline/history library uses it

Re: [Freeswitch-users] Recomended VOIP Providers?

Providers? Please have a look at the wiki http://wiki.freeswitch.org/wiki/SIP_Provider_Examples In which country do you need a provider? So far I got every provider I tried (5) to work with freeswitch; even with FS behind NAT. Andy Ayers schrieb: Hi, Can anyone recommend any good VOIP providers

Re: [Freeswitch-users] Behavior of deflect

Woof! On Thu, 20 Nov 2008 19:06:09 -0500, Anthony Minessale [EMAIL PROTECTED] wrote: I wonder if sofia is not matching the notify to the dialog so we are not associating it with the channel. I know for a fact sofia tears it down for us when it gets the notify. do you have x-lite/eyebeam?

Re: [Freeswitch-users] Behavior of deflect

Woof! On Fri, 21 Nov 2008 14:49:35 -0500, Anthony Minessale [EMAIL PROTECTED] wrote: can you try uuid_deflect again on latest, Done. I see we give up to easy, on 180 instead of final response in sipfrag the one i tested didnt send 180 so i forgot about the possibility. post a full console

Re: [Freeswitch-users] Behavior of deflect

Woof! Anthony wrote: I asked because I was wondering if you could test the scenario I described to compare what happens to a call from x-lite being deflected since I know for a fact it was working. Woof wrote: FS still doesn't hangup on the original call after the REFER is completed. I

Re: [Freeswitch-users] Behavior of deflect

Woof! Thanks for the changes! On Wed, 19 Nov 2008 21:09:19 -0500, Anthony Minessale [EMAIL PROTECTED] wrote: try latest code and see how that works. FreeSWITCH Version 1.0.trunk (10481) No difference with just deflect--the call does not clear when the REFER is completed, nor are there any

Re: [Freeswitch-users] Behavior of deflect

Woof! On Thu, 20 Nov 2008 15:14:52 -0500, Anthony Minessale [EMAIL PROTECTED] wrote: how were you calling the deflect the way that had no change? every time i tried it sofia has taken down the channel once it completed. Hmm... dialplan: extension name=IVR condition

[Freeswitch-users] Behavior of deflect

Woof! When using deflect on an inbound answered call, I notice that the FS channel stays connected as long as the original call exists, and does not send a BYE to the original call even after receiving a NOTFIY with a final response fragment. In addition, while FS gets those NOTIFY's, I

[Freeswitch-users] Recomended VOIP Providers?

. regards Andy ___ Freeswitch-users mailing list 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

Re: [Freeswitch-users] Freeswitch Java Socket Interface API

Woof! On Mon, 03 Nov 2008 08:55:11 -0500, Klaus Teller [EMAIL PROTECTED] wrote: Hi Folks, Just to let you know that we are working on a library for connecting to the Freeswitch via the socket interface. We plan to release it under LGPL as soon as it's somewhat robust. You may be

[Freeswitch-users] Status of 1.02?

Woof! Is there any updated projection of when 1.0.2 will be released? Last I can find mention of it is over a month ago (http://freeswitch.org/node/143) Thanks, --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] What happend to variable_* in socket_outbound?

Woof! On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris [EMAIL PROTECTED] wrote: They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event. Nope. Initial event. No variable_* are reported. Using netcat: Connection

Re: [Freeswitch-users] What happend to variable_* in socket_outbound?

Woof! On Wed, 29 Oct 2008 14:31:36 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: update and try again ;) Anthony, you are just too fast! Mike was saying one thing, and you checked in the change it while I was replying to him! Updating now. --Woof!

[Freeswitch-users] What happend to variable_* in socket_outbound?

Woof! I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back? --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] mod_conference: Any way to pass conference PIN in the URI?

Woof! On Mon, 13 Oct 2008 11:02:49 -0400, Kristian Kielhofner [EMAIL PROTECTED] wrote: Instead of modifying the URI, why not attach your own header: X-conf-pin: 1234 I haven't done it yet, but it might be as simple as: action application=conference data=confname+${sip_X-conf-pin} I

[Freeswitch-users] VoipTalk NAT

. Many thanks Andy ___ Freeswitch-users mailing list 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] Assertion and core dump on FreeSWITCH shutdown

Woof! We've been seeing core dumps on FreeSWITCH shutdown. I just updated to today's revision (r9526) and still see them: freeswitch: src/switch_core_hash.c:59: switch_core_hash_destroy: Assertion `hash != ((void *)0) *hash != ((void *)0)' failed. Core was generated by

Re: [Freeswitch-users] Assertion and core dump on FreeSWITCH shutdown

Woof! On Thu, 11 Sep 2008 15:57:26 -0400, Michael Jerris [EMAIL PROTECTED] wrote: Fixed in svn r9527, thanks for the report. Verified. Thanks. --Woof! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org

Re: [Freeswitch-users] mod_perl buid errors on Solaris

Woof! On Mon, 18 Aug 2008 14:25:45 -0400, Bruce McAlister [EMAIL PROTECTED] wrote: Does anyone have any idea's on the error listed below? /opt/SUNWspro/bin/cc -w -DMULTIPLICITY -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED_PERL

Re: [Freeswitch-users] speak with dtmf collect

Wof! On Fri, 25 Jul 2008 13:09:45 -0400, Boris Krivonog [EMAIL PROTECTED] wrote: Hi all! I'm using event sockets to remotely drive freeswitch. Is there a read like functionality for speak that can be used with event sockets, for example: One way to handle DTMF collection with event

Re: [Freeswitch-users] Stupid question of the week: how to generate silence and still detect DTMF

Woof! On Tue, 08 Jul 2008 19:13:24 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: try now =D Ahh, now there is joy! Can I trouble you to also make tone_stream://%(150, 0, 0) (or something similar) also work? I can forsee a need for injecting short pauses into a sequence of prompts, and

Re: [Freeswitch-users] Stupid question of the week: how to generate silence and still detect DTMF

Woof! On Wed, 09 Jul 2008 12:49:51 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: you can try 1,1 instead of 0,0 which is still silent Actually, it isn't. It is quite audible on several phones I've played with (most likely due to the uLaw companding adding a step to each voltage, and

Re: [Freeswitch-users] Stupid question of the week: how to generate silence and still detect DTMF

Woof! On Tue, 08 Jul 2008 13:27:40 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: Try latest trunk, I added support for what you want. Thanks. I just tried it. DTMF event reporting now works...but I cannot break out of the sleep command once it is started. --Woof!

Re: [Freeswitch-users] Stupid question of the week: how to generate silence and still detect DTMF

Woof! On Tue, 08 Jul 2008 15:27:02 -0400, Anthony Minessale [EMAIL PROTECTED] wrote: update and try again. Alas, no joy. Break still won't wake it up. svn version 8933 --Woof! ___ Freeswitch-users mailing list

[Freeswitch-users] Proxy authentication on REFER

Woof! Using an event socket on an answered parked call, I am trying to use deflect to transfer the call off of FreeSwitch to a SIP destination, like this: sendmsg call-command: execute execute-app-name: deflect execute-app-arg: sip:[EMAIL PROTECTED] Alas, our proxy will challenge the REFER

Re: [Freeswitch-users] Delayed DTMF events during gentones

Woof! On Fri, 23 May 2008 17:13:59 -0400, Brian West [EMAIL PROTECTED] wrote: How are you calling the socket application for the outbound event socket connection? If you put it in async you should get those events. I've tried it various ways, but this is the one I'm currently using:

Re: [Freeswitch-users] Delayed DTMF events during gentones

Woof! On Fri, 23 May 2008 20:02:46 -0400, Brian West [EMAIL PROTECTED] wrote: Please update to at least rev 8564 and try again. Will do next Tuesday. Thanks for the quick fixes. Enjoy the long weekend (for those of you who are US based, at least!) --Woof!