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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ajit M Kallingal Sent: Wednesday, July 30, 2003 7:08 PM To: [EMAIL PROTECTED] 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: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users digest, Vol 1 #935 - 14 msgs > Send Asterisk-Users mailing list submissions to > [EMAIL PROTECTED] > > 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 > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:49:56 -0400 > Reply-To: [EMAIL PROTECTED] > > 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 > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 2 > Date: Wed, 30 Jul 2003 15:08:53 -0500 (CDT) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 16:12:59 -0400 > Reply-To: [EMAIL PROTECTED] > > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Todd > Lieberman > Sent: Wednesday, July 30, 2003 3:50 PM > To: [EMAIL PROTECTED] > 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/gid script > is writable by world., referer: > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > Mannheimer > Sent: Wednesday, July 30, 2003 3:23 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > > > Thanks! > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > Lesher > Sent: July 30, 2003 4:06 PM > To: [EMAIL PROTECTED] > 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 > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > --__--__-- > > Message: 4 > From: Tilghman Lesher <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] voicemail file access problems > Date: Wed, 30 Jul 2003 15:18:20 -0500 > Reply-To: [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > > 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 > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 6 > Date: Wed, 30 Jul 2003 16:33:21 -0400 (EDT) > From: Patrick <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] voicemail file access problems > Reply-To: [EMAIL PROTECTED] > > > 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 it worked. > > > > you still need to make sure nobody has read/write permission on > > /var/spool/asterisk/vm/$MBOX > > > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Todd > > Lieberman > > Sent: Wednesday, July 30, 2003 3:50 PM > > To: [EMAIL PROTECTED] > > 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/gid script > > is writable by world., referer: > > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] Behalf Of Paulo > > Mannheimer > > Sent: Wednesday, July 30, 2003 3:23 PM > > To: [EMAIL PROTECTED] > > Subject: RE: [Asterisk-Users] voicemail file access problems > > > > > > Thanks! > > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman > > Lesher > > Sent: July 30, 2003 4:06 PM > > To: [EMAIL PROTECTED] > > 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 > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 7 > Date: Wed, 30 Jul 2003 15:42:43 -0500 (CDT) > From: Brian West <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > 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 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 > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > --__--__-- > > Message: 8 > Date: Wed, 30 Jul 2003 16:48:42 -0400 (EDT) > From: Patrick <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn > Reply-To: [EMAIL PROTECTED] > > > 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 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 > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > --__--__-- > > Message: 9 > From: "Darren Smith" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Date: Wed, 30 Jul 2003 21:55:41 +0100 > Organization: Game Digital Ltd > Subject: [Asterisk-Users] X100P and incoming Context + CDR? > Reply-To: [EMAIL PROTECTED] > > Hi folks > > I have a X100P in my home asterisk box and I have it setup as a default context 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, asterisk answers, plays them > the contents of my mp3 partition whilst my SIP phone rings, I answer and talk to the poor > soul about my useless taste in music. > > However, in the CDR records it says the destination number is 's', is there anyway I can > change this? > > Someone mentioned there was a app_setDNIS function at some point but it seems to have > vanished again, or can i do it directly in asterisk/zaptel? > > Regards > > Darren Smith > Game Digital Ltd > > --__--__-- > > Message: 10 > From: "Kyle Hagan" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] CVS Problem? > Date: Wed, 30 Jul 2003 14:01:48 -0700 > Reply-To: [EMAIL PROTECTED] > > 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: [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] href=3D"mailto:[EMAIL PROTECTED]">Kyle = > Hagan</A>=20 > </DIV> > <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20 > [EMAIL PROTECTED] > = > href=3D"mailto:[EMAIL PROTECTED]">[EMAIL PROTECTED] > 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 <[EMAIL PROTECTED]> > To: [EMAIL PROTECTED] > Organization: > Date: 30 Jul 2003 16:22:11 -0500 > Reply-To: [EMAIL PROTECTED] > > 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 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 > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > 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: [EMAIL PROTECTED] > From: "Carlos Crembil" <[EMAIL PROTECTED]> > Date: Wed, 30 Jul 2003 17:25:23 -0300 > Subject: [Asterisk-Users] %unsuscribe > Reply-To: [EMAIL PROTECTED] > > > %unsuscribe > > > > --__--__-- > > Message: 13 > Date: Wed, 30 Jul 2003 23:31:58 +0200 (CEST) > From: Siggi Langauf <[EMAIL PROTECTED]> > To: Asterisk user list <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] SetCIDName > Reply-To: [EMAIL PROTECTED] > > 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" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] X-Lite and Call transfer using Asterisk > Date: Wed, 30 Jul 2003 22:44:54 +0100 > Reply-To: [EMAIL PROTECTED] > > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Steven J. > Sobol > Sent: 30 July 2003 20:20 > To: [EMAIL PROTECTED] > 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 * [EMAIL PROTECTED] > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > --__--__-- > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users