[Asterisk-Users] MAC OS X Panther :?

Mon Jul 19 10:02:29 MST 2004

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..

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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>
> 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 ?!?!
>
> _______________________________________________
> 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
>
>
>
> --__--__--
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> End of Asterisk-Users Digest
>