[Asterisk-Users] SCO/Linux concerns

Joe Antkowiak joe at jsci.net
Wed Jul 30 20:21:04 MST 2003 

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 at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Ajit M Kallingal
Sent: Wednesday, July 30, 2003 7:08 PM
To: asterisk-users at 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 at lists.digium.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, July 30, 2003 3:05 PM
Subject: Asterisk-Users digest, Vol 1 #935 - 14 msgs


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>
> 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 at tlsolutions.net>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] voicemail file access problems
> Date: Wed, 30 Jul 2003 15:49:56 -0400
> Reply-To: asterisk-users at 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/gid
script
> is writable by world., referer:
> http://asterisk.weichertrents.com/cgi-bin/vmail.cgi
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Paulo
> Mannheimer
> Sent: Wednesday, July 30, 2003 3:23 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] voicemail file access problems
>
>
> Thanks!
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Tilghman
> Lesher
> Sent: July 30, 2003 4:06 PM
> To: asterisk-users at 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 at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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 at bkw.org>
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] sip -> h323 -> ptsn
> Reply-To: asterisk-users at 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 at tlsolutions.net>
> To: <asterisk-users at lists.digium.com>
> Subject: RE: [Asterisk-Users] voicemail file access problems
> Date: Wed, 30 Jul 2003 16:12:59 -0400
> Reply-To: asterisk-users at 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 at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Todd
> Lieberman
> Sent: Wednesday, July 30, 2003 3:50 PM
> To: asterisk-users at 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/gid
script
> is writable by world., referer:
> http://asterisk.weichertrents.com/cgi-bin/vmail.cgi
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Paulo
> Mannheimer
> Sent: Wednesday, July 30, 2003 3:23 PM
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] voicemail file access problems
>
>
> Thanks!
>
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Tilghman
> Lesher
> Sent: July 30, 2003 4:06 PM
> To: asterisk-users at 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 at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --__--__--
>
> Message: 4
> From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] voicemail file access problems
> Date: Wed, 30 Jul 2003 15:18:20 -0500
> Reply-To: asterisk-users at 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 at sip2.dmv.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn
> Reply-To: asterisk-users at 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 at 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 at sip2.dmv.com>
> To: asterisk-users at lists.digium.com
> Subject: RE: [Asterisk-Users] voicemail file access problems
> Reply-To: asterisk-users at 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 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 at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Todd
> > Lieberman
> > Sent: Wednesday, July 30, 2003 3:50 PM
> > To: asterisk-users at 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/gid
script
> > is writable by world., referer:
> > http://asterisk.weichertrents.com/cgi-bin/vmail.cgi
> >
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Paulo
> > Mannheimer
> > Sent: Wednesday, July 30, 2003 3:23 PM
> > To: asterisk-users at lists.digium.com
> > Subject: RE: [Asterisk-Users] voicemail file access problems
> >
> >
> > Thanks!
> >
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Tilghman
> > Lesher
> > Sent: July 30, 2003 4:06 PM
> > To: asterisk-users at 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 at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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 at bkw.org>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn
> Reply-To: asterisk-users at 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 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 at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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 at sip2.dmv.com>
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn
> Reply-To: asterisk-users at 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 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 at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --__--__--
>
> Message: 9
> From: "Darren Smith" <data at barrysworld.com>
> To: <asterisk-users at 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 at lists.digium.com
>
> 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" <khagan at nuvoinc.com>
> To: <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] CVS Problem?
> Date: Wed, 30 Jul 2003 14:01:48 -0700
> Reply-To: asterisk-users at 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 at 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>&nbsp;</DIV>
> <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</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 at nuvoinc.com href=3D"mailto:khagan at nuvoinc.com">Kyle =
> Hagan</A>=20
>   </DIV>
>   <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
>   title=3Dasterisk-users at lists.digium.com=20
>   =
> href=3D"mailto:asterisk-users at lists.digium.com">asterisk-users at 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>&nbsp;</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>&nbsp;</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>&nbsp;</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 at fnords.org>
> To: asterisk-users at lists.digium.com
> Organization:
> Date: 30 Jul 2003 16:22:11 -0500
> Reply-To: asterisk-users at 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 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 at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at 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 at lists.digium.com
> From: "Carlos Crembil" <ccrembil at openware.biz>
> Date: Wed, 30 Jul 2003 17:25:23 -0300
> Subject: [Asterisk-Users] %unsuscribe
> Reply-To: asterisk-users at lists.digium.com
>
>
> %unsuscribe
>
>
>
> --__--__--
>
> Message: 13
> Date: Wed, 30 Jul 2003 23:31:58 +0200 (CEST)
> From: Siggi Langauf <langausd at swt.uni-stuttgart.de>
> To: Asterisk user list <asterisk-users at lists.digium.com>
> Subject: Re: [Asterisk-Users] SetCIDName
> Reply-To: asterisk-users at 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 at easynet.co.uk>
> To: <asterisk-users at 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 at 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 at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Steven J.
> Sobol
> Sent: 30 July 2003 20:20
> To: asterisk-users at 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 at JustThe.net
>
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