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: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, July 19, 2004 12:25 PM > Subject: Asterisk-Users digest, Vol 1 #4598 - 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: 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? ([EMAIL PROTECTED]) > > 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? ([EMAIL PROTECTED]) > > 14. RE: STILL NO AUDIO (Sebastian Nocetti) > > > > --__--__-- > > > > Message: 1 > > Date: Mon, 19 Jul 2004 18:24:39 +0300 > > From: Michael Manousos <[EMAIL PROTECTED]> > > Organization: inAccess Networks > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] STILL NO AUDIO > > Reply-To: [EMAIL PROTECTED] > > > > > > Why don't you use asterisk-oh323? > > > > Michael. > > > > Sebastian Nocetti wrote: > > > I WANT TO USE G729, I HAVE TO USE IT... > > > > > > -----Mensaje original----- > > > De: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling > > > Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. > > > Para: [EMAIL PROTECTED] > > > 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 <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config > > Reply-To: [EMAIL PROTECTED] > > > > 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 <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] ZyXEL 2000W > > Reply-To: [EMAIL PROTECTED] > > > > On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager <[EMAIL PROTECTED]> > 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: [EMAIL PROTECTED] > > From: "Chris A. Icide" <[EMAIL PROTECTED]> > > Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no > > Reply-To: [EMAIL PROTECTED] > > > > 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 <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Organization: BTEL Consulting > > Date: Mon, 19 Jul 2004 10:27:22 -0500 > > Reply-To: [EMAIL PROTECTED] > > > > 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 <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] STILL NO AUDIO > > Date: Mon, 19 Jul 2004 17:32:14 +0200 > > Reply-To: [EMAIL PROTECTED] > > > > > 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" <[EMAIL PROTECTED]> > > To: "'[EMAIL PROTECTED]'" <[EMAIL PROTECTED]> > > Subject: RE: [Asterisk-Users] Mac OS X installer for Asterisk > > Date: Mon, 19 Jul 2004 11:28:24 -0400 > > Reply-To: [EMAIL PROTECTED] > > > > 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:[EMAIL PROTECTED] > > 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 > > [EMAIL PROTECTED] > > 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:[EMAIL PROTECTED] > > REV:20040709T130909Z > > END:VCARD > > > > ------_=_NextPart_000_01C46DA5.08080030-- > > > > --__--__-- > > > > Message: 8 > > Date: Mon, 19 Jul 2004 10:39:53 -0500 > > From: [EMAIL PROTECTED] > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] PhoneGaim? > > Reply-To: [EMAIL PROTECTED] > > > > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] BroadVoice problems? > > Date: Mon, 19 Jul 2004 08:43:07 -0700 > > Reply-To: [EMAIL PROTECTED] > > > > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 > > > [EMAIL PROTECTED] > > > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Subject: RE: [Asterisk-Users] STILL NO AUDIO > > Date: Mon, 19 Jul 2004 12:51:49 -0300 > > Reply-To: [EMAIL PROTECTED] > > > > Testing both... > > > > -----Mensaje original----- > > De: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] En nombre de Michael > Manousos > > Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m. > > Para: [EMAIL PROTECTED] > > 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: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] En nombre de Eric > > > Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. > > > Para: [EMAIL PROTECTED] > > > 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 > > [EMAIL PROTECTED] > > 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 <[EMAIL PROTECTED]> > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config > > Reply-To: [EMAIL PROTECTED] > > > > On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley <[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Date: Mon, 19 Jul 2004 17:04:58 +0100 > > Subject: [Asterisk-Users] IP Phone recommendation > > Reply-To: [EMAIL PROTECTED] > > > > 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: [EMAIL PROTECTED] <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? > > Cc: > > Reply-To: [EMAIL PROTECTED] > > > > Look out for 3c17205 switches from 3com and read the QOS thread posting > here at the moment. > > > > P > > > > > -----Original Message----- > > > From: Scott Laird [mailto:[EMAIL PROTECTED] > > > Sent: Monday, July 19, 2004, 7:58 AM > > > To: '[EMAIL PROTECTED]' <[EMAIL PROTECTED]> > > > 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 > > > [EMAIL PROTECTED] > > > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Subject: RE: [Asterisk-Users] STILL NO AUDIO > > Date: Mon, 19 Jul 2004 13:08:10 -0300 > > Reply-To: [EMAIL PROTECTED] > > > > What kind of problem? > > > > All works OK except that config.... > > > > -----Mensaje original----- > > De: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] En nombre de Holger Schurig > > Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m. > > Para: [EMAIL PROTECTED] > > 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 > > [EMAIL PROTECTED] > > 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 > > [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 > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
