[Asterisk-Users] MAC OS X Panther :?

[Josh Krueger]

On Mon, 2004-07-19 at 12:02, Francisco Perez-Landaeta wrote:
> Just wondering if anyone has tried MAC OS X and panther.
> I will like to do SIP to H323, not sure if this will be possible on the MAC
> because of the Libraries PWlib and OPenh32 for Linux..

Is this for running asterisk on OS X? or for a soft phone? If it's a
soft phone, then there shouldn't be a problem.

> 
> Just curious..
> 
> Anyway, anyone has an easy guide (step by step) to setup oh323 with
> asterisk. I saw a guide but i am not very savy on linux.
> thanks,
> Francisco
> 
> ----- Original Message ----- 
> From: <asterisk-users-request at lists.digium.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Monday, July 19, 2004 12:25 PM
> Subject: Asterisk-Users digest, Vol 1 #4598 - 14 msgs
> 
> 
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> > or, via email, send a message with subject or body 'help' to
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> > than "Re: Contents of Asterisk-Users digest..."
> >
> >
> > Today's Topics:
> >
> >    1. Re: STILL NO AUDIO (Michael Manousos)
> >    2. Re: TDM400P Internal Extenion Config (Nick Cobley)
> >    3. Re: ZyXEL 2000W (Jason Williams)
> >    4. Channel banks, voicemail, and immediate=no (Chris A. Icide)
> >    5. RE: STILL NO AUDIO (Eric Wieling)
> >    6. Re: STILL NO AUDIO (Holger Schurig)
> >    7. RE: Mac OS X installer for Asterisk (Wallingford, Ted)
> >    8. Re: PhoneGaim? (creslin at digium.com)
> >    9. Re: BroadVoice problems? (Chris Shaw)
> >   10. RE: STILL NO AUDIO (Sebastian Nocetti)
> >   11. Re: TDM400P Internal Extenion Config (Jason Williams)
> >   12. IP Phone recommendation (Yiannis Costopoulos)
> >   13. Re: Cheap PoE switches/injectors? (asteriskstuff at ziplip.com)
> >   14. RE: STILL NO AUDIO (Sebastian Nocetti)
> >
> > --__--__--
> >
> > Message: 1
> > Date: Mon, 19 Jul 2004 18:24:39 +0300
> > From: Michael Manousos <manousos at inaccessnetworks.com>
> > Organization: inAccess Networks
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] STILL NO AUDIO
> > Reply-To: asterisk-users at lists.digium.com
> >
> >
> > Why don't you use asterisk-oh323?
> >
> > Michael.
> >
> > Sebastian Nocetti wrote:
> > > I WANT TO USE G729, I HAVE TO USE IT...
> > >
> > > -----Mensaje original-----
> > > De: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] En nombre de Eric Wieling
> > > Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m.
> > > Para: asterisk-users at lists.digium.com
> > > Asunto: Re: [Asterisk-Users] STILL NO AUDIO
> > >
> > > I suspect it will be solved when you put disallow=all and allow=ulaw in
> > > sip.conf and h323.conf (and NO OTHER ALLOW= LINES)
> > >
> > > On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote:
> > >
> > >>I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when
> > >>connected, NOTHING....
> > >>
> > >>
> > >>
> > >>It happened in both: SIP -> CHAN_H323 and CHAN_H323 -> SIP...
> > >>
> > >>
> > >>
> > >>when it will be solved?
> >
> >
> > --__--__--
> >
> > Message: 2
> > Date: Mon, 19 Jul 2004 23:26:06 +0800
> > From: Nick Cobley <info at nvworld.net>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config
> > Reply-To: asterisk-users at lists.digium.com
> >
> > Thanks Steve,
> >
> > The SIP handsets are working find as I can make calls to other handsets
> > as well as receive incoming calls via the FXO module. So all is good
> there.
> >
> > Cheers
> > Nick
> >
> > Steven Critchfield wrote:
> >
> > >On Mon, 2004-07-19 at 07:13, Nick Cobley wrote:
> > >
> > >
> > >
> > >>If I dial the extension I just get a 404 error on the phone
> > >>(Grandstream), but no errors at all on the console. I am using
> > >>CVS-HEAD-07/14/04.  Here is a snippet of what I have in the various
> > >>config files.
> > >>
> > >>
> > >
> > >Welcome to SIP. Dialtone is local to your phone and is not dependent on
> > >proper config. Hope that helps put you on the correct step to fix that
> > >problem.
> > >
> > >
> >
> >
> > --__--__--
> >
> > Message: 3
> > Date: Mon, 19 Jul 2004 16:26:26 +0100
> > From: Jason Williams <jas.williams at gmail.com>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] ZyXEL 2000W
> > Reply-To: asterisk-users at lists.digium.com
> >
> > On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager <andrew at rwts.com.au>
> wrote:
> > > Does anyone have the call hold feature working? If you do... how did
> > > you make it work? The instructions say to press the left button to
> > > place the call on hold, and the right button to take it off - except
> > > when I am in a call, these keys have no effect.
> > >
> > > I've tried teh 000c firmware, the 000e firmware and the Pulver 0011
> > > firmware - but none work, so I'm wondering if this feature just simply
> > > isn't implemented, or if there is likely to be something wrong with my
> > > asterisk config.
> >
> > No it does not work, you need to use # transfer which will mean you
> > will not be able to dial # into ivr's.
> >
> > Search on wiki for # transfer
> >
> > Regards
> >
> >
> > Jason
> >
> > --__--__--
> >
> > Message: 4
> > Date: Mon, 19 Jul 2004 08:26:32 -0700
> > To: asterisk-users at lists.digium.com
> > From: "Chris A. Icide" <chris at netgeeks.net>
> > Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no
> > Reply-To: asterisk-users at lists.digium.com
> >
> > When using a channel bank for analog handsets, you have a couple options
> in
> > the way you handle transactions involving the analog handsets and
> origination.
> >
> > With immediate set to no, it appears to me that soon as a digit is pressed
> > after going off-hook, the single digit is taken and processed against the
> > context that the channel is associated with from the configuration in
> > zapata.conf.
> >
> > With immediate set to yes, the extension s in the channel's context is
> > processed.
> >
> > As far as I know, the method of handling channel bank based analog
> handsets
> > is to use immediate=yes and then have extension s put the phone directly
> > into a DISA command with no-password and a context for processing the
> > entered calls.
> >
> > I have also tried in the past setting immediate=no, parsing off the first
> > digit and sending the call into separate contexts (see example below)
> >
> > example with immediate=yes
> >
> > exten => s,1,DISA,no-password|internal
> >
> >
> > example with immediate=no
> >
> > exten => 9,1,DISA,no-password|pstn-gateway
> >
> >
> > In the first case, the problem I have is this:  If I place the handset
> > directly into DISA, how can I get stuttertone MWI indication?
> >
> > If I use the second method, in many cases, there is NO dialtone provided
> to
> > the phone until after a dtmf entry is recieved.  This I suspect is a
> > channel bank issue because it seems to work on some banks, and not on
> others.
> >
> >
> > Given the use of channel banks as a method to allow large number of analog
> > phones to access an asterisk system, is there any way (or perhaps any
> > interest in developing a method) to actually treat analog handsets on a
> > channel bank like any other UA?  In other words, why not have a method
> > besides the two above so that I can stick the phones into a context (which
> > understands it's for handling analog phones on a channel bank) that
> > actually provides dial tone, and accepts dtmf until a match to the context
> > extensions is found?  In other words, with immediate=no, I'd like to see
> > asterisk not jump on the first dtmf and try to match (going to i, if no
> > match exists), but actually wait for as many dtmf's as required to match
> an
> > extension in the context (e.g. exten => _1NXXNXXXXXX waits for 10 digits
> if
> > dtmf 1 is the first digit).
> >
> >
> > On a different track, am I doing something wrong above?  For people who
> > have configured channel banks for use with asterisk, have you found a
> > 'perfect' configuration that you prefer to use?
> >
> > -Chris
> >
> >
> > --__--__--
> >
> > Message: 5
> > Subject: RE: [Asterisk-Users] STILL NO AUDIO
> > From: Eric Wieling <eric at fnords.org>
> > To: asterisk-users at lists.digium.com
> > Organization: BTEL Consulting
> > Date: Mon, 19 Jul 2004 10:27:22 -0500
> > Reply-To: asterisk-users at lists.digium.com
> >
> > On Mon, 2004-07-19 at 10:00, Sebastian Nocetti wrote:
> > > I WANT TO USE G729, I HAVE TO USE IT...
> >
> > Not while testing you don't.  Once you get it working with ULAW ONLY
> > then see if you can get it working with G729.
> > -- 
> > Useful Asterisk Docs (BOOKMARK THEM!):
> > http://www.digium.com/index.php?menu=documentation (look at the
> > "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and
> > http://www.fnords.org/~eric/asterisk/ (my site) and
> > http://asteriskdocs.org/
> >
> >
> > --__--__--
> >
> > Message: 6
> > From: Holger Schurig <hs4233 at mail.mn-solutions.de>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] STILL NO AUDIO
> > Date: Mon, 19 Jul 2004 17:32:14 +0200
> > Reply-To: asterisk-users at lists.digium.com
> >
> > > I WANT TO USE G729, I HAVE TO USE IT...
> >
> > When you have no FW and no NAT, then you seem to be inside your local
> > network. In this case you shouldn't really care ?!?!
> >
> >
> > --__--__--
> >
> > Message: 7
> > From: "Wallingford, Ted" <ted at indexc.com>
> > To: "'asterisk-users at lists.digium.com'" <asterisk-users at lists.digium.com>
> > Subject: RE: [Asterisk-Users] Mac OS X installer for Asterisk
> > Date: Mon, 19 Jul 2004 11:28:24 -0400
> > Reply-To: asterisk-users at lists.digium.com
> >
> > This message is in MIME format. Since your mail reader does not understand
> > this format, some or all of this message may not be legible.
> >
> > ------_=_NextPart_000_01C46DA5.08080030
> > Content-Type: text/plain
> >
> > Benjamin,
> >
> > Is this package intended to mirror the directory structure of the linux
> > builds? If so, I may have an issue: While /var/lib/asterisk is properly in
> > place after running the installer, /usr/sbin/asterisk is not. I'm running
> on
> > OS X 10.3.4 and downloaded the package on Sunday afternoon, if that's any
> > help. Did I miss something?
> >
> > Thanks,
> > Ted Wallingford
> >
> >
> > -----Original Message-----
> > From: Sunrise Ltd [mailto:stsltdtyo at yahoo.co.jp]
> > Sent: Saturday, July 17, 2004 2:09 PM
> > To: astusr
> > Subject: [Asterisk-Users] Mac OS X installer for Asterisk
> >
> >
> > Hi
> >
> > I have created a Mac OS X installer package for installing Asterisk on OSX
> > ver 10.2 and 10.3
> >
> > Anyone who'd like to give this a try, please download the installer
> package
> > from here ...
> >
> > http://www.astmasters.net/stuff/Asterisk.pkg.tgz
> >
> > to install Asterisk on OSX just double click the package
> > file.
> >
> > please send any feedback to benjamin (at) sunrise (dash)
> > tel (dot) com
> >
> > NOTE: this is a fairly old build but it's rock solid. We
> > have run it on OSX Server 10.2.8 since October last year
> > and it's been going like a Swiss clockwork. Rich Murphey
> > has promised to fix the Makefile for the most recent CVS
> > so it will build on OSX again. Once this is done, we'll
> > make another installer package for the new version.
> >
> > Also, I am still working on extending the install package
> > so that users can choose whether or not they want to
> > install the sources. Anybody interested in this, please
> > bare with me a few more days.
> >
> > regards
> > benjamin
> >
> > --
> > Sunrise Telephone Systems Ltd
> > 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku,
> > Tokyo, Japan
> >
> >
> > __________________________________________________
> > Do You Yahoo!?
> > http://bb.yahoo.co.jp/
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > ------_=_NextPart_000_01C46DA5.08080030
> > Content-Type: application/octet-stream;
> > name="Wallingford, Ted.vcf"
> > Content-Disposition: attachment;
> > filename="Wallingford, Ted.vcf"
> >
> > BEGIN:VCARD
> > VERSION:2.1
> > N:Wallingford;Ted
> > FN:Wallingford, Ted
> > EMAIL;PREF;INTERNET:ted at indexc.com
> > REV:20040709T130909Z
> > END:VCARD
> >
> > ------_=_NextPart_000_01C46DA5.08080030--
> >
> > --__--__--
> >
> > Message: 8
> > Date: Mon, 19 Jul 2004 10:39:53 -0500
> > From: creslin at digium.com
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] PhoneGaim?
> > Reply-To: asterisk-users at lists.digium.com
> >
> > On Sun, Jul 18, 2004 at 10:12:20AM -0500, Chris Howard wrote:
> > > I say on slashdot that the Linspire guys have released PhoneGaim.
> > > PhoneGaim is Gaim with SIP added on.  Anyone want to add IAX2 as
> > > well...
> >
> > I'm writing a plugin for gaim right now that does iax2 on my off time.
> > I haven't had much time to work on it lately, but I'm right now at kind
> > of a decision point for what hooks will be in gaim to interface it.
> > Maybe like a iaxtel/* protocol plugin.  I'm still speculating about
> > details though.  I've got most of the lower stuff done now.
> >
> > Matthew Fredrickson
> >
> > --__--__--
> >
> > Message: 9
> > From: "Chris Shaw" <chriss at watertech.com>
> > To: <asterisk-users at lists.digium.com>
> > Subject: Re: [Asterisk-Users] BroadVoice problems?
> > Date: Mon, 19 Jul 2004 08:43:07 -0700
> > Reply-To: asterisk-users at lists.digium.com
> >
> > Now that you mention it, yes... it seems that SIP isn't being passed from
> > their PSTN gateway to the rest of their network... It's ringing, but
> there's
> > no acknowledgement in * that anything's going on...
> >
> > ----- Original Message -----
> > From: "Chris Tooley" <ctooley at ntrc.net>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Monday, July 19, 2004 8:19 AM
> > Subject: [Asterisk-Users] BroadVoice problems?
> >
> >
> > > Anyone else having problems with inbound Broadvoice this morning?
> > > --
> > > Chris Tooley / Network and Development Services
> > > Networking Technologies Resource Center, LLC (NTRC)
> > > 8650 Spicewood Springs Road, Suite 105
> > > Austin TX 78759
> > > 512-250-8985 / Fax 512-250-5909
> > > www.ntrc.net / www.ntrcstore.com
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --__--__--
> >
> > Message: 10
> > From: "Sebastian Nocetti" <snocetti at fibertel.com.ar>
> > To: <asterisk-users at lists.digium.com>
> > Subject: RE: [Asterisk-Users] STILL NO AUDIO
> > Date: Mon, 19 Jul 2004 12:51:49 -0300
> > Reply-To: asterisk-users at lists.digium.com
> >
> > Testing both...
> >
> > -----Mensaje original-----
> > De: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] En nombre de Michael
> Manousos
> > Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m.
> > Para: asterisk-users at lists.digium.com
> > Asunto: Re: [Asterisk-Users] STILL NO AUDIO
> >
> >
> > Why don't you use asterisk-oh323?
> >
> > Michael.
> >
> > Sebastian Nocetti wrote:
> > > I WANT TO USE G729, I HAVE TO USE IT...
> > >
> > > -----Mensaje original-----
> > > De: asterisk-users-admin at lists.digium.com
> > > [mailto:asterisk-users-admin at lists.digium.com] En nombre de Eric
> > > Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m.
> > > Para: asterisk-users at lists.digium.com
> > > Asunto: Re: [Asterisk-Users] STILL NO AUDIO
> > >
> > > I suspect it will be solved when you put disallow=all and allow=ulaw
> > > in sip.conf and h323.conf (and NO OTHER ALLOW= LINES)
> > >
> > > On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote:
> > >
> > >>I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when
> > >>connected, NOTHING....
> > >>
> > >>
> > >>
> > >>It happened in both: SIP -> CHAN_H323 and CHAN_H323 -> SIP...
> > >>
> > >>
> > >>
> > >>when it will be solved?
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > --__--__--
> >
> > Message: 11
> > Date: Mon, 19 Jul 2004 16:57:48 +0100
> > From: Jason Williams <jas.williams at gmail.com>
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config
> > Reply-To: asterisk-users at lists.digium.com
> >
> > On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley <info at nvworld.net> wrote:
> > > Hopefully someone here can save my sanity. I have been trying to solve
> > > this problem for days now, but just cant put my finger on it. Im new to
> > > * so I have probably done something stupid!
> > Only a config issue I'm sure
> >
> > > [sip]
> > > exten => 301,1,Dial(SIP/Nick,20,tr)
> > > exten => 302,1,Dial(SIP/Sharon,20,tr)
> > > exten => 1000,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
> > > exten => 302,2,VoiceMail,u302
> > > exten => 301,2,VoiceMail,u301
> > > exten => 1000,2,VoiceMail,u9999
> > > exten => 1000,102,VoiceMail,b9999
> > > exten => 1001,1,Ringing
> > > exten => 1001,2,Wait(2)
> > > exten => 1001,3,VoicemailMain
> > > include => outgoing
> > add here
> > include => internal  ; allow sip to dial 310
> >
> > > [incoming]
> > > exten => s,1,Dial(SIP/Nick&SIP/Sharon,20,tr)
> > >
> > > [outgoing]
> > > exten => _7.,1,Dial(IAX2/login:passwd at XXX.XXX.XXX>XXX/${EXTEN:1})
> > > exten => 5.,1,Dial,Zap/1/${EXTEN:1}
> > >
> > > [9103]
> > > exten => 21060,1,Dial(SIP/Nick)
> > > exten => 21062,1,Dial(SIP/Sharon)
> > >
> > > [internal]
> > > exten => 310,1,Dial,Zap/2
> > include => sip ; allow internal to dial sip phone
> > >
> >
> > Try those changes and see how you get on
> >
> >
> > Jason
> >
> > --__--__--
> >
> > Message: 12
> > From: "Yiannis Costopoulos" <yiannis at w2ns.com>
> > To: <asterisk-users at lists.digium.com>
> > Date: Mon, 19 Jul 2004 17:04:58 +0100
> > Subject: [Asterisk-Users] IP Phone recommendation
> > Reply-To: asterisk-users at lists.digium.com
> >
> > Hi,
> >
> > I am looking for some affordable IP Phones. Any experiences with the
> > SipToneII by ipDialog?
> >
> > What about soft phones? Any recommendations there (for Windoze and Linux)?
> >
> > Thanks,
> > Yiannis
> >
> >
> > --__--__--
> >
> > Message: 13
> > Date: Mon, 19 Jul 2004 09:03:49 -0700 (PDT)
> > From: asteriskstuff at ziplip.com <asteriskstuff at ziplip.com>
> > To: <asterisk-users at lists.digium.com>
> > Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors?
> > Cc:
> > Reply-To: asterisk-users at lists.digium.com
> >
> > Look out for 3c17205 switches from 3com and read the QOS thread posting
> here at the moment.
> >
> > P
> >
> > > -----Original Message-----
> > > From: Scott Laird [mailto:scott at sigkill.org]
> > > Sent: Monday, July 19, 2004, 7:58 AM
> > > To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com>
> > > Subject: [Asterisk-Users] Cheap PoE switches/injectors?
> > >
> > > I'm trying to spec out hardware for a new office, and I'd like to
> > > include power over Ethernet as an option.  I've seen a handful of PoE
> > > injectors around $1000 for 24 ports and a couple switches up around
> > > $2500 for 24 ports.  Are there any cheaper options, short of buying a
> > > boatload of 1-port injectors off of ebay?  I don't really need more
> > > then 24 ports of PoE out of 48 total ports, so one of CIsco's big PoE
> > > switches is complete overkill.  This is for a startup, where cheap is
> > > important.
> > >
> > > Thanks.
> > >
> > >
> > > Scott
> > >
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --__--__--
> >
> > Message: 14
> > From: "Sebastian Nocetti" <snocetti at fibertel.com.ar>
> > To: <asterisk-users at lists.digium.com>
> > Subject: RE: [Asterisk-Users] STILL NO AUDIO
> > Date: Mon, 19 Jul 2004 13:08:10 -0300
> > Reply-To: asterisk-users at lists.digium.com
> >
> > What kind of problem?
> >
> > All works OK except that config....
> >
> > -----Mensaje original-----
> > De: asterisk-users-admin at lists.digium.com
> > [mailto:asterisk-users-admin at lists.digium.com] En nombre de Holger Schurig
> > Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m.
> > Para: asterisk-users at lists.digium.com
> > Asunto: Re: [Asterisk-Users] STILL NO AUDIO
> >
> > > I WANT TO USE G729, I HAVE TO USE IT...
> >
> > When you have no FW and no NAT, then you seem to be inside your local
> > network. In this case you shouldn't really care ?!?!
> >
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