Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 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: voicemail file access problems (Todd Lieberman) > 2. sip -> h323 -> ptsn (Brian West) > 3. RE: voicemail file access problems (Todd Lieberman) > 4. Re: voicemail file access problems (Tilghman Lesher) > 5. Re: sip -> h323 -> ptsn (Patrick) > 6. RE: voicemail file access problems (Patrick) > 7. Re: sip -> h323 -> ptsn (Brian West) > 8. Re: sip -> h323 -> ptsn (Patrick) > 9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: asterisk-users@lists.digium.com > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > _______________________________________________ > 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: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <brian@bkw.org> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > > --__--__-- > > Message: 3 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: asterisk-users@lists.digium.com > > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > _______________________________________________ > 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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > From: Tilghman Lesher <tilghman@mail.jeffandtilghman.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:18:20 -0500 > Reply-To: asterisk-users@lists.digium.com > > On Wednesday 30 July 2003 02:49 pm, Todd Lieberman wrote: > > I did the chown and now I get > > > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] > > Setuid/gid script is writable by world., referer: > > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > chmod o-w vmail.cgi > > btw, 'man chmod' helps. Blindly executing commands as root > that you received on a public mailing list is usually not a fine > idea. > > -Tilghman > > > --__--__-- > > Message: 5 > Date: Wed, 30 Jul 2003 16:26:24 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have this setup: > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > Sip phones are setup for out of band dtmf > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > the sip phones to the ptsn via the h323 gateway? > > > > bkw > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 6 > Date: Wed, 30 Jul 2003 16:33:21 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > Reply-To: asterisk-users@lists.digium.com > > > Did it work after you left a new voice mail message? > > I was looking into the source code to fix it so that the euid was set to > nobody, create the file and then change it back to uid 0, but that didn't > work. Or, maybe change the file mode was 770 with the group set so that > the webserver could modify the file so I wouldn't have to run a suid .cgi > script. > > Patrick > > On Wed, 30 Jul 2003, Todd Lieberman wrote: > > > I fixed my own problem. I had just did chmod 755 vmail.cgi and itworked.> > > > you still need to make sure nobody has read/write permission on > > /var/spool/asterisk/vm/$MBOX > > > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd > > Lieberman > > Sent: Wednesday, July 30, 2003 3:50 PM > > To: asterisk-users@lists.digium.com > > Subject: RE: [Asterisk-Users] voicemail file access problems > > > > > > I did the chown and now I get > > > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> > is writable by world., referer: > > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > > Mannheimer > > Sent: Wednesday, July 30, 2003 3:23 PM > > To: asterisk-users@lists.digium.com > > Subject: RE: [Asterisk-Users] voicemail file access problems > > > > > > Thanks! > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > > Lesher > > Sent: July 30, 2003 4:06 PM > > To: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] voicemail file access problems > > > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > > Hi folks, > > > > > > I'm having problems accessing my voicemail files through the web > > > interface. > > > > > > I remember that this was discussed on the list, and it seems to be > > > a permission problem, but I couldn't find any answer by searching > > > the archives. > > > > > > Any hint? > > > > chown root vmail.cgi > > chmod u+s vmail.cgi > > > > -Tilghman > > > > _______________________________________________ > > 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 > > > > _______________________________________________ > > 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: 7 > Date: Wed, 30 Jul 2003 15:42:43 -0500 (CDT) > From: Brian West <brian@bkw.org> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > I have done that but I think its the Asterisk => MC3810 via h323 thats > causing that. Does anyone have an example on how i can dump sip to and > from the MC3810 to my asterisk server? > > bkw > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > _______________________________________________ > > > 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: 8 > Date: Wed, 30 Jul 2003 16:48:42 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > > Try setting dtmf-relay h245-alphanumeric in the MC3810 dial-peer. > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have done that but I think its the Asterisk => MC3810 via h323 thats > > causing that. Does anyone have an example on how i can dump sip to and > > from the MC3810 to my asterisk server? > > > > bkw > > > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > > > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > > > I have this setup: > > > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > > > Sip phones are setup for out of band dtmf > > > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > > the sip phones to the ptsn via the h323 gateway? > > > > > > > > bkw > > > > _______________________________________________ > > > > 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 > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 9 > From: "Darren Smith" <data@barrysworld.com> > To: <asterisk-users@lists.digium.com> > Date: Wed, 30 Jul 2003 21:55:41 +0100 > Organization: Game Digital Ltd > Subject: [Asterisk-Users] X100P and incoming Context + CDR? > Reply-To: asterisk-users@lists.digium.com > > Hi folks > > I have a X100P in my home asterisk box and I have it setup as a defaultcontext of> 'incoming-pstn' > > in my extensions.conf i have > > [incoming-pstn] > exten => s,1,Goto(incoming,01225<myofficenumber>,1) > > then: > > [incoming] > > exten => 01225<myofficenumber>,1,Answer > exten => 01225<myofficenumber>,2,Dial(SIP/data|m) > etc etc > > Anywho back to the plot. > > It all works wonderful, someone dials my home office line, asteriskanswers, plays them> the contents of my mp3 partition whilst my SIP phone rings, I answer andtalk to the poor> soul about my useless taste in music. > > However, in the CDR records it says the destination number is 's', isthere anyway I can> change this? > > Someone mentioned there was a app_setDNIS function at some point but itseems to have> vanished again, or can i do it directly in asterisk/zaptel? > > Regards > > Darren Smith > Game Digital Ltd > > --__--__-- > > Message: 10 > From: "Kyle Hagan" <khagan@nuvoinc.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] CVS Problem? > Date: Wed, 30 Jul 2003 14:01:48 -0700 > Reply-To: asterisk-users@lists.digium.com > > This is a multi-part message in MIME format. > > ------=_NextPart_000_0050_01C356A3.1E8422E0 > Content-Type: text/plain; > charset="iso-8859-1" > Content-Transfer-Encoding: quoted-printable > > I figured it out. I had a file called CVS in the directory and it > freaked out.. > > > Kyle > ----- Original Message -----=20 > From: Kyle Hagan=20 > To: asterisk-users@lists.digium.com=20 > Sent: Wednesday, July 30, 2003 9:23 AM > Subject: [Asterisk-Users] CVS Problem? > > > Since yesterday i get the following message when downloading anything > from the CVS. > > cvs [checkout aborted]: reading CVS/Tag: Not a directory > > Is it a problem on my end or digium? I havnt changed anything on my > end. > > Kyle > > > ------=_NextPart_000_0050_01C356A3.1E8422E0 > Content-Type: text/html; > charset="iso-8859-1" > Content-Transfer-Encoding: quoted-printable > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> > <HTML><HEAD> > <META http-equiv=3DContent-Type content=3D"text/html; > charset=3Diso-8859-1"> > <META content=3D"MSHTML 6.00.2800.1170" name=3DGENERATOR> > <STYLE></STYLE> > </HEAD> > <BODY bgColor=3D#ffffff> > <DIV><FONT face=3DArial size=3D2>I figured it out. I had a file called > CVS in the=20 > directory and it freaked out..</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Kyle</FONT></DIV> > <BLOCKQUOTE dir=3Dltr=20 > style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; > BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px"> > <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV> > <DIV=20 > style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: > black"><B>From:</B>=20 > <A title=3Dkhagan@nuvoinc.com href=3D"mailto:khagan@nuvoinc.com">Kyle > Hagan</A>=20 > </DIV> > <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20 > title=3Dasterisk-users@lists.digium.com=20 > > href=3D"mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digi> um.com</A>=20 > </DIV> > <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, July 30, 2003 > 9:23=20 > AM</DIV> > <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] CVS=20 > Problem?</DIV> > <DIV><BR></DIV> > <DIV><FONT face=3DArial size=3D2>Since yesterday i get the following > message when=20 > downloading anything from the CVS.</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>cvs [checkout aborted]: reading > CVS/Tag: Not a=20 > directory</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Is it a problem on my end or digium? > I havnt=20 > changed anything on my end.</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Kyle</DIV> > <DIV><BR></DIV></BLOCKQUOTE></FONT></BODY></HTML> > > ------=_NextPart_000_0050_01C356A3.1E8422E0-- > > > --__--__-- > > Message: 11 > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > From: Eric Wieling <eric@fnords.org> > To: asterisk-users@lists.digium.com > Organization: > Date: 30 Jul 2003 16:22:11 -0500 > Reply-To: asterisk-users@lists.digium.com > > That only works if you are using the G711 (ulaw/alaw) codecs. Other > codecs distort inband DTMF. > > On Wed, 2003-07-30 at 15:26, Patrick wrote: > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > _______________________________________________ > > > 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 > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > > --__--__-- > > Message: 12 > To: asterisk-users@lists.digium.com > From: "Carlos Crembil" <ccrembil@openware.biz> > Date: Wed, 30 Jul 2003 17:25:23 -0300 > Subject: [Asterisk-Users] %unsuscribe > Reply-To: asterisk-users@lists.digium.com > > > %unsuscribe > > > > --__--__-- > > Message: 13 > Date: Wed, 30 Jul 2003 23:31:58 +0200 (CEST) > From: Siggi Langauf <langausd@swt.uni-stuttgart.de> > To: Asterisk user list <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] SetCIDName > Reply-To: asterisk-users@lists.digium.com > > On Wed, 30 Jul 2003, Jeremy McNamara wrote: > > > Because H.323 doesn't have a specific 'feature' of caller*id. > > However, it does seem to have > - calling party number > - calling party name > - display string > > and at least the last one seems to be set to whatever SetCallerID() tells > it to be if you're using chan_oh323 from inaccessnetworks, so that string > is displayed on the called party's phone... > > > --__--__-- > > Message: 14 > From: "Stuart Hirst" <shirst@easynet.co.uk> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk > Date: Wed, 30 Jul 2003 22:44:54 +0100 > Reply-To: asterisk-users@lists.digium.com > > I have the same with the transfer issue but also when I call between > X-Lite and a SNOM 200 there is no audio but if I call between X-Lite and > a Budgetone 102 all is OK. > > Stuart > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven J. > Sobol > Sent: 30 July 2003 20:20 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk > > > On Wed, 30 Jul 2003, Brian West wrote: > > > Same here. Same build. > > <AOL> > > -- > JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & > Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 > Steve Sobol, Proprietor > 888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.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 > > > End of Asterisk-Users Digest >
What's your concern with it? If any of SCO code made it into GNU stuff, it will be removed and rewritten in a short time anyway... -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ajit M Kallingal Sent: Wednesday, July 30, 2003 7:08 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SCO/Linux concerns Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 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: voicemail file access problems (Todd Lieberman) > 2. sip -> h323 -> ptsn (Brian West) > 3. RE: voicemail file access problems (Todd Lieberman) > 4. Re: voicemail file access problems (Tilghman Lesher) > 5. Re: sip -> h323 -> ptsn (Patrick) > 6. RE: voicemail file access problems (Patrick) > 7. Re: sip -> h323 -> ptsn (Brian West) > 8. Re: sip -> h323 -> ptsn (Patrick) > 9. X100P and incoming Context + CDR? (Darren Smith) > 10. Re: CVS Problem? (Kyle Hagan) > 11. Re: sip -> h323 -> ptsn (Eric Wieling) > 12. %unsuscribe (Carlos Crembil) > 13. Re: SetCIDName (Siggi Langauf) > 14. RE: X-Lite and Call transfer using Asterisk (Stuart Hirst) > > --__--__-- > > Message: 1 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: asterisk-users@lists.digium.com > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > _______________________________________________ > 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: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <brian@bkw.org> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > I have this setup: > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > Sip phones are setup for out of band dtmf > > but the h323 gateway is inband. Is their a way to pass the digits from > the sip phones to the ptsn via the h323 gateway? > > bkw > > --__--__-- > > Message: 3 > From: "Todd Lieberman" <todd@tlsolutions.net> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: asterisk-users@lists.digium.com > > I fixed my own problem. I had just did chmod 755 vmail.cgi and it worked. > > you still need to make sure nobody has read/write permission on > /var/spool/asterisk/vm/$MBOX > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > I did the chown and now I get > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > Hi folks, > > > > I'm having problems accessing my voicemail files through the web > > interface. > > > > I remember that this was discussed on the list, and it seems to be > > a permission problem, but I couldn't find any answer by searching > > the archives. > > > > Any hint? > > chown root vmail.cgi > chmod u+s vmail.cgi > > -Tilghman > > _______________________________________________ > 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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > From: Tilghman Lesher <tilghman@mail.jeffandtilghman.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:18:20 -0500 > Reply-To: asterisk-users@lists.digium.com > > On Wednesday 30 July 2003 02:49 pm, Todd Lieberman wrote: > > I did the chown and now I get > > > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] > > Setuid/gid script is writable by world., referer: > > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > chmod o-w vmail.cgi > > btw, 'man chmod' helps. Blindly executing commands as root > that you received on a public mailing list is usually not a fine > idea. > > -Tilghman > > > --__--__-- > > Message: 5 > Date: Wed, 30 Jul 2003 16:26:24 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > > I have the same setup, and in the sip.conf file I set the dtmfmode=inband > for each endpoint defined and my Cisco ATA-186s and 7960 phones all work. > > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have this setup: > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > Sip phones are setup for out of band dtmf > > > > but the h323 gateway is inband. Is their a way to pass the digits from > > the sip phones to the ptsn via the h323 gateway? > > > > bkw > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 6 > Date: Wed, 30 Jul 2003 16:33:21 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] voicemail file access problems > Reply-To: asterisk-users@lists.digium.com > > > Did it work after you left a new voice mail message? > > I was looking into the source code to fix it so that the euid was set to > nobody, create the file and then change it back to uid 0, but that didn't > work. Or, maybe change the file mode was 770 with the group set so that > the webserver could modify the file so I wouldn't have to run a suid .cgi > script. > > Patrick > > On Wed, 30 Jul 2003, Todd Lieberman wrote: > > > I fixed my own problem. I had just did chmod 755 vmail.cgi and itworked.> > > > you still need to make sure nobody has read/write permission on > > /var/spool/asterisk/vm/$MBOX > > > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Todd > > Lieberman > > Sent: Wednesday, July 30, 2003 3:50 PM > > To: asterisk-users@lists.digium.com > > Subject: RE: [Asterisk-Users] voicemail file access problems > > > > > > I did the chown and now I get > > > > [Wed Jul 30 15:51:11 2003] [error] [client 216.183.124.45] Setuid/gidscript> > is writable by world., referer: > > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Paulo > > Mannheimer > > Sent: Wednesday, July 30, 2003 3:23 PM > > To: asterisk-users@lists.digium.com > > Subject: RE: [Asterisk-Users] voicemail file access problems > > > > > > Thanks! > > > > -----Original Message----- > > From: asterisk-users-admin@lists.digium.com > > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Tilghman > > Lesher > > Sent: July 30, 2003 4:06 PM > > To: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] voicemail file access problems > > > > On Wednesday 30 July 2003 01:41 pm, Paulo Mannheimer wrote: > > > Hi folks, > > > > > > I'm having problems accessing my voicemail files through the web > > > interface. > > > > > > I remember that this was discussed on the list, and it seems to be > > > a permission problem, but I couldn't find any answer by searching > > > the archives. > > > > > > Any hint? > > > > chown root vmail.cgi > > chmod u+s vmail.cgi > > > > -Tilghman > > > > _______________________________________________ > > 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 > > > > _______________________________________________ > > 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: 7 > Date: Wed, 30 Jul 2003 15:42:43 -0500 (CDT) > From: Brian West <brian@bkw.org> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > I have done that but I think its the Asterisk => MC3810 via h323 thats > causing that. Does anyone have an example on how i can dump sip to and > from the MC3810 to my asterisk server? > > bkw > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > _______________________________________________ > > > 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: 8 > Date: Wed, 30 Jul 2003 16:48:42 -0400 (EDT) > From: Patrick <patrick@sip2.dmv.com> > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: asterisk-users@lists.digium.com > > > Try setting dtmf-relay h245-alphanumeric in the MC3810 dial-peer. > > On Wed, 30 Jul 2003, Brian West wrote: > > > I have done that but I think its the Asterisk => MC3810 via h323 thats > > causing that. Does anyone have an example on how i can dump sip to and > > from the MC3810 to my asterisk server? > > > > bkw > > > > On Wed, 30 Jul 2003, Patrick wrote: > > > > > > > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > > > I have this setup: > > > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > > > Sip phones are setup for out of band dtmf > > > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > > the sip phones to the ptsn via the h323 gateway? > > > > > > > > bkw > > > > _______________________________________________ > > > > 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 > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 9 > From: "Darren Smith" <data@barrysworld.com> > To: <asterisk-users@lists.digium.com> > Date: Wed, 30 Jul 2003 21:55:41 +0100 > Organization: Game Digital Ltd > Subject: [Asterisk-Users] X100P and incoming Context + CDR? > Reply-To: asterisk-users@lists.digium.com > > Hi folks > > I have a X100P in my home asterisk box and I have it setup as a defaultcontext of> 'incoming-pstn' > > in my extensions.conf i have > > [incoming-pstn] > exten => s,1,Goto(incoming,01225<myofficenumber>,1) > > then: > > [incoming] > > exten => 01225<myofficenumber>,1,Answer > exten => 01225<myofficenumber>,2,Dial(SIP/data|m) > etc etc > > Anywho back to the plot. > > It all works wonderful, someone dials my home office line, asteriskanswers, plays them> the contents of my mp3 partition whilst my SIP phone rings, I answer andtalk to the poor> soul about my useless taste in music. > > However, in the CDR records it says the destination number is 's', isthere anyway I can> change this? > > Someone mentioned there was a app_setDNIS function at some point but itseems to have> vanished again, or can i do it directly in asterisk/zaptel? > > Regards > > Darren Smith > Game Digital Ltd > > --__--__-- > > Message: 10 > From: "Kyle Hagan" <khagan@nuvoinc.com> > To: <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] CVS Problem? > Date: Wed, 30 Jul 2003 14:01:48 -0700 > Reply-To: asterisk-users@lists.digium.com > > This is a multi-part message in MIME format. > > ------=_NextPart_000_0050_01C356A3.1E8422E0 > Content-Type: text/plain; > charset="iso-8859-1" > Content-Transfer-Encoding: quoted-printable > > I figured it out. I had a file called CVS in the directory and it > freaked out.. > > > Kyle > ----- Original Message -----=20 > From: Kyle Hagan=20 > To: asterisk-users@lists.digium.com=20 > Sent: Wednesday, July 30, 2003 9:23 AM > Subject: [Asterisk-Users] CVS Problem? > > > Since yesterday i get the following message when downloading anything > from the CVS. > > cvs [checkout aborted]: reading CVS/Tag: Not a directory > > Is it a problem on my end or digium? I havnt changed anything on my > end. > > Kyle > > > ------=_NextPart_000_0050_01C356A3.1E8422E0 > Content-Type: text/html; > charset="iso-8859-1" > Content-Transfer-Encoding: quoted-printable > > <!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> > <HTML><HEAD> > <META http-equiv=3DContent-Type content=3D"text/html; > charset=3Diso-8859-1"> > <META content=3D"MSHTML 6.00.2800.1170" name=3DGENERATOR> > <STYLE></STYLE> > </HEAD> > <BODY bgColor=3D#ffffff> > <DIV><FONT face=3DArial size=3D2>I figured it out. I had a file called > CVS in the=20 > directory and it freaked out..</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Kyle</FONT></DIV> > <BLOCKQUOTE dir=3Dltr=20 > style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; > BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px"> > <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV> > <DIV=20 > style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: > black"><B>From:</B>=20 > <A title=3Dkhagan@nuvoinc.com href=3D"mailto:khagan@nuvoinc.com">Kyle > Hagan</A>=20 > </DIV> > <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20 > title=3Dasterisk-users@lists.digium.com=20 > > href=3D"mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digi> um.com</A>=20 > </DIV> > <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, July 30, 2003 > 9:23=20 > AM</DIV> > <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [Asterisk-Users] CVS=20 > Problem?</DIV> > <DIV><BR></DIV> > <DIV><FONT face=3DArial size=3D2>Since yesterday i get the following > message when=20 > downloading anything from the CVS.</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>cvs [checkout aborted]: reading > CVS/Tag: Not a=20 > directory</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Is it a problem on my end or digium? > I havnt=20 > changed anything on my end.</FONT></DIV> > <DIV><FONT face=3DArial size=3D2></FONT> </DIV> > <DIV><FONT face=3DArial size=3D2>Kyle</DIV> > <DIV><BR></DIV></BLOCKQUOTE></FONT></BODY></HTML> > > ------=_NextPart_000_0050_01C356A3.1E8422E0-- > > > --__--__-- > > Message: 11 > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > From: Eric Wieling <eric@fnords.org> > To: asterisk-users@lists.digium.com > Organization: > Date: 30 Jul 2003 16:22:11 -0500 > Reply-To: asterisk-users@lists.digium.com > > That only works if you are using the G711 (ulaw/alaw) codecs. Other > codecs distort inband DTMF. > > On Wed, 2003-07-30 at 15:26, Patrick wrote: > > I have the same setup, and in the sip.conf file I set thedtmfmode=inband> > for each endpoint defined and my Cisco ATA-186s and 7960 phones allwork.> > > > > > On Wed, 30 Jul 2003, Brian West wrote: > > > > > I have this setup: > > > > > > Sip Phones -> Asterisk -> h323 gateway -> ptsn > > > > > > Sip phones are setup for out of band dtmf > > > > > > but the h323 gateway is inband. Is their a way to pass the digitsfrom> > > the sip phones to the ptsn via the h323 gateway? > > > > > > bkw > > > _______________________________________________ > > > 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 > -- > BTEL Consulting > 850-484-4535 x2111 (Office) > 504-595-3916 x2111 (Experimental) > 877-552-0838 (Backup Phone) > > > --__--__-- > > Message: 12 > To: asterisk-users@lists.digium.com > From: "Carlos Crembil" <ccrembil@openware.biz> > Date: Wed, 30 Jul 2003 17:25:23 -0300 > Subject: [Asterisk-Users] %unsuscribe > Reply-To: asterisk-users@lists.digium.com > > > %unsuscribe > > > > --__--__-- > > Message: 13 > Date: Wed, 30 Jul 2003 23:31:58 +0200 (CEST) > From: Siggi Langauf <langausd@swt.uni-stuttgart.de> > To: Asterisk user list <asterisk-users@lists.digium.com> > Subject: Re: [Asterisk-Users] SetCIDName > Reply-To: asterisk-users@lists.digium.com > > On Wed, 30 Jul 2003, Jeremy McNamara wrote: > > > Because H.323 doesn't have a specific 'feature' of caller*id. > > However, it does seem to have > - calling party number > - calling party name > - display string > > and at least the last one seems to be set to whatever SetCallerID() tells > it to be if you're using chan_oh323 from inaccessnetworks, so that string > is displayed on the called party's phone... > > > --__--__-- > > Message: 14 > From: "Stuart Hirst" <shirst@easynet.co.uk> > To: <asterisk-users@lists.digium.com> > Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk > Date: Wed, 30 Jul 2003 22:44:54 +0100 > Reply-To: asterisk-users@lists.digium.com > > I have the same with the transfer issue but also when I call between > X-Lite and a SNOM 200 there is no audio but if I call between X-Lite and > a Budgetone 102 all is OK. > > Stuart > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven J. > Sobol > Sent: 30 July 2003 20:20 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk > > > On Wed, 30 Jul 2003, Brian West wrote: > > > Same here. Same build. > > <AOL> > > -- > JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & > Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 > Steve Sobol, Proprietor > 888.480.4NET (4638) * 248.724.4NET * sjsobol@JustThe.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 > > > End of Asterisk-Users Digest >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users
On Wed, 2003-07-30 at 18:07, Ajit M Kallingal wrote:> Hello > Since I am getting a bit concerned about the SCO vs IBM issue, I was > wondering if can I can setup Asterisk on FreeBSD is it supported ? > Are drivers for Digium cards available on FreeBSD ?If you are worried about it, you really should look into what happened during the lawsuit between Bellcore/USL and UC berkely. Basically they removed all the offending parts from BSD and then rewrote them from scratch. That is why there is a 4.4 lite BSD, it was they after settlement release. It wasn't a full unix anymore, but it had enough to start from again. From that core, you get the *BSD systems. The worst that will happen is a judge will deem certain parts as infringing and order them removed. At which time the functionality will be replaced by originally written code. No problems. So far the only chance of finding infringing code from SCO to linux was donated by a caldera employee. And do remember Caldera purchased the Unix code and then changed names to the SCO group. It isn't even the people who used to write and maintain it. The real fun is when the judge goes in and finds the infringing code in SCO's linux compatibility code that caldera employees have rumored about. If the GPL is enforceable, then all of the historical unix code would fall under GPL then and the lawsuit will blow away in a poof of logic. -- Steven Critchfield <critch@basesys.com>
On Wednesday 30 July 2003 07:07 pm, Ajit M Kallingal wrote:> Hello > Since I am getting a bit concerned about the SCO vs IBM issue, I was > wondering if can I can setup Asterisk on FreeBSD is it supported ? > Are drivers for Digium cards available on FreeBSD ? > > Thanks > AjitNot really answering you but, the odds are indeed very slim, it will take long time to hash through court. 1) By the time any bad ruling would have been made the Linux community would have a time (as trial moves on) to remove/rewrite any offending code. 2) If a ruling is done for SCO they will pursue BIG multi billion dollar corp first. Why? Because they want to get as much money as possible. 3) Finally, there's no chance that some 10 million plus users who have in good faith used Linux would be found guilty by a court. Solving this non existent problem by not using Linux is a waste of time. Not because being proactive is bad but because it is not even close to be a REAL situation at this point. We should all use Linux happily until the scene changes. At this point we even have governments all around the world using it. Try other O/S because you want to not because of SCO. SCO is using media, who live to scare people, to make more money by selling millions of stock at new hightend stock prices. This is a fact. I for one have filed a complaint with the SEC over this apparent scam. Germany have stopped them from continuing allegations without evidence. Get on with Asterisk and Linux... -- Steve ______________________________________ This sig is pending approval
On Wednesday 30 July 2003 19:07, Ajit M Kallingal wrote:> I am getting a bit concerned about the SCO vs IBM issueYou may wish to read a recent position paper by Eben Moglen. http://osdl.org/osdlpress/2003_07_31_beaverton.html