Re: [asterisk-users] Zapata.conf
On Sat, Jan 31, 2009 at 12:00:21PM -0800, Steve Edwards wrote: > On Sat, 31 Jan 2009, William Muriithi wrote: > > > I am however to find zapata.conf. I have searched the whole system for > > the name zapata, and none seem to materialize. I am currently interested > > to know where it lives. Is it /etc/zapata.conf or > > /etc/asterisk/zapata.conf? > > zapata.conf lives in /etc/asterisk/. > > (For future reference, "sudo find / | grep -i zapata.conf" may help.) find / -name zapata.conf Or better: sudo updatedb # if actually needed locate zapata.conf > > It can be created using genzaptelconf which can be found in > zaptel/xpp/utils/. zapconf is now preffered, BTW. Though it defaults to create a partial zapata.conf file. > > It can be created using "make samples" from your Asterisk source > directory, but this may clobber other parts of your configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
On Sat, Jan 31, 2009 at 11:25:40PM +, Edwin Quijada wrote:
>
>
>
>
> >
> > You say "Connected" but do not specify in what fashions you are
> > connecting. That piece of info will be the solution. I have done
> > this many times in many fashions.
> >
> Sorry for my bad english but I dont understand what info you need to know.
>
> You can ask me for anything that u need.
>
> Conection by H323 protocol
> Using The NuFone Network's Open H.323 driver configuration
> Pwlib for asterisk \
> If u need something else, just ask me.!
I'm not really an H323 expert, but some obvious questions come to mind:
1. Did this work before on the same system? If so, when did it stop
working?
2. Any chance that this is an over-zelious firewall in the middle? Can
you make sure you're not blocking any packets with iptables?
3. What are the relevant configurations?
4. What does happen? What do you see in the CLI trace? Is there any
h323 protocol-level debug? ('h323 debug' or something similar)? What do
you see there?
BTW: JerJer is no longer the maintainer of chan_h323 .
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Quiet 24 port POE gig switch
I can find FANLESS 24 port PoE 10/100, or FANLESS 24 port non-POE 10/100/1000 I guess I'll just have to wait for newer chips..till then dropping down to 10/100 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: January 31, 2009 10:02 PM To: Asterisk Users List Subject: Re: [asterisk-users] Quiet 24 port POE gig switch Importance: High OCG Technical Support wrote: > A little off topic but > > > > I need to put a 24 port Gig PoE switch into a small office - no computer > room / rack etc. All CAT5 terminates near the owners desk (smart huh?). > > > > I want to put a PoE switch in place, with 24 ports and Gig speed. > Everyone I've researched so far is LOUD... Chances of finding a PoE switch that is quiet out of the box is about as good as finding a government 'worker'. It's kind of an oxymoron. Of the switches I've used, the Linksys/Cisco line was the loudest. Dlink's were quieter, but still not something you'd want sitting next to a desk. About the only fanless PoE switches I've seen are the smaller Netgear's, but they are not Gigabit. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
OCG Technical Support wrote: > A little off topic but > > > > I need to put a 24 port Gig PoE switch into a small office – no computer > room / rack etc. All CAT5 terminates near the owners desk (smart huh?). > > > > I want to put a PoE switch in place, with 24 ports and Gig speed. > Everyone I’ve researched so far is LOUD... Chances of finding a PoE switch that is quiet out of the box is about as good as finding a government 'worker'. It's kind of an oxymoron. Of the switches I've used, the Linksys/Cisco line was the loudest. Dlink's were quieter, but still not something you'd want sitting next to a desk. About the only fanless PoE switches I've seen are the smaller Netgear's, but they are not Gigabit. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
You will have a hard time finding a 24 port POE without fans - too high of a power density. Do you really need 24 ports? perhaps a 12x12 otherwise multiple 8 or 12 port models may work Do let us know if you find a 24 port without fans. On Jan 31, 2009, at 2:06 PM, Claus Herwig wrote: > Hello, > >> I need to put a 24 port Gig PoE switch into a small office – no >> computer >> room / rack etc. All CAT5 terminates near the owners desk (smart >> huh?). > > I had a similar problem some days ago. 24-port GBit Switch in the > middle > of a classroom... > > I ended up with a kind of "semi-loud" setup: I bought a 3Com 3CBLSG > switch (this is without PoE, but there is a PoE version of it). There > are two quite noisy 40x40x10 fans inside. I replaced the two with one > 40x40x20 ebmPapst silent fan (model 412/2, 18dB), left the other fan > offline and mounted the whole thing vertically so that convection > supports the remaining fan. I tried with two silent fans (enough space > inside), but this still was too noisy. > > Some measurement indicates the cooling is sufficient this way. But > understand that I've no long term data, as I installed this setup just > two weeks ago. And of course your warranty is void ;-) > > Greets, > Claus > > -- > CHECON EDV-Consulting und Redaktion > Claus Herwig * Barer Straße 70 * 80799 München > +49 89 27826981 * Fax 27826982 * c.her...@checon.de > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
hi
one way audio maybe is a codec problem.
check it.
and the avaya has the third party h323 licence rigth?
David
2009/1/31 Steve Totaro
> I was going to suggest T1 but you seemed set on H323. T1 is T1 you
> may have some configuration issues but it will work.
>
> On Sat, Jan 31, 2009 at 7:08 PM, Edwin Quijada
> wrote:
> >
> > Well, thks anyway :)
> > Maybe in another ocasion with T1 trunk :)
> >
> >
> >
> > *---*
> > *-Edwin Quijada
> > *-Developer DataBase
> > *-JQ Microsistemas
> > *-809-849-8087
> >
> > * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
> comun"
> > *---*
> >
> >
> >
> >
> >
> >
> >
> >> Date: Sat, 31 Jan 2009 18:58:09 -0500
> >> From: stot...@first-notification.com
> >> To: asterisk-users@lists.digium.com
> >> Subject: Re: [asterisk-users] Avaya and Asterisk sound one-way
> >>
> >> On Sat, Jan 31, 2009 at 6:42 PM, Steve Totaro
> >> wrote:
> >>> On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada
> >>> wrote:
>
>
>
>
> >
> > You say "Connected" but do not specify in what fashions you are
> > connecting. That piece of info will be the solution. I have done
> > this many times in many fashions.
> >
> Sorry for my bad english but I dont understand what info you need to
> know.
>
> You can ask me for anything that u need.
>
> Conection by H323 protocol
> Using The NuFone Network's Open H.323 driver configuration
> Pwlib for asterisk \
> If u need something else, just ask me.!
>
>
>
>
>
> *---*
> *-Edwin Quijada
> *-Developer DataBase
> *-JQ Microsistemas
> *-809-849-8087
>
> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de
> lo comun"
> *---*
>
>
>
>
>
> _
> Stay up to date on your PC, the Web, and your mobile phone with
> Windows Live
> http://clk.atdmt.com/MRT/go/119462413/direct/01/
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> >>>
> >>> H323 is a dog in Asterisk, JerJer can probaby help you out ot this one.
> >>>
> >>> --
> >>> Thanks,
> >>> Steve Totaro
> >>> +18887771888 (Toll Free)
> >>> +12409381212 (Cell)
> >>> +12024369784 (Skype)
> >>>
> >>
> >> Bye the way, JerJer is the man at NuFone. I don't use H323 but I
> >> believe there are other H323 implementations other than JerJer's you
> >> might want to try. JerJer was never a help to me and purposely ripped
> >> me off for ~$40, this was years ago, I prepaid and the service was
> >> down with no tech support, so I asked for a refund that I never got.
> >> It is in the archives somewhere.
> >>
> >> JerJer/NuFone=bad at least at the beginning. I would look elsewhere.
> >>
> >> --
> >> Thanks,
> >> Steve Totaro
> >> +18887771888 (Toll Free)
> >> +12409381212 (Cell)
> >> +12024369784 (Skype)
> >>
> >> ___
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > _
> > Get 5 GB of storage with Windows Live Hotmail.
> >
> http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
> ___
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Re: [asterisk-users] Avaya and Asterisk sound one-way
I was going to suggest T1 but you seemed set on H323. T1 is T1 you may have some configuration issues but it will work. On Sat, Jan 31, 2009 at 7:08 PM, Edwin Quijada wrote: > > Well, thks anyway :) > Maybe in another ocasion with T1 trunk :) > > > > *---* > *-Edwin Quijada > *-Developer DataBase > *-JQ Microsistemas > *-809-849-8087 > > * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo > comun" > *---* > > > > > > > >> Date: Sat, 31 Jan 2009 18:58:09 -0500 >> From: stot...@first-notification.com >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Avaya and Asterisk sound one-way >> >> On Sat, Jan 31, 2009 at 6:42 PM, Steve Totaro >> wrote: >>> On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada >>> wrote: > > You say "Connected" but do not specify in what fashions you are > connecting. That piece of info will be the solution. I have done > this many times in many fashions. > Sorry for my bad english but I dont understand what info you need to know. You can ask me for anything that u need. Conection by H323 protocol Using The NuFone Network's Open H.323 driver configuration Pwlib for asterisk \ If u need something else, just ask me.! *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun" *---* _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> H323 is a dog in Asterisk, JerJer can probaby help you out ot this one. >>> >>> -- >>> Thanks, >>> Steve Totaro >>> +18887771888 (Toll Free) >>> +12409381212 (Cell) >>> +12024369784 (Skype) >>> >> >> Bye the way, JerJer is the man at NuFone. I don't use H323 but I >> believe there are other H323 implementations other than JerJer's you >> might want to try. JerJer was never a help to me and purposely ripped >> me off for ~$40, this was years ago, I prepaid and the service was >> down with no tech support, so I asked for a refund that I never got. >> It is in the archives somewhere. >> >> JerJer/NuFone=bad at least at the beginning. I would look elsewhere. >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> +12024369784 (Skype) >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > _ > Get 5 GB of storage with Windows Live Hotmail. > http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
Well, thks anyway :) Maybe in another ocasion with T1 trunk :) *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun" *---* > Date: Sat, 31 Jan 2009 18:58:09 -0500 > From: stot...@first-notification.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Avaya and Asterisk sound one-way > > On Sat, Jan 31, 2009 at 6:42 PM, Steve Totaro > wrote: >> On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada >> wrote: >>> >>> >>> >>> You say "Connected" but do not specify in what fashions you are connecting. That piece of info will be the solution. I have done this many times in many fashions. >>> Sorry for my bad english but I dont understand what info you need to know. >>> >>> You can ask me for anything that u need. >>> >>> Conection by H323 protocol >>> Using The NuFone Network's Open H.323 driver configuration >>> Pwlib for asterisk \ >>> If u need something else, just ask me.! >>> >>> >>> >>> >>> >>> *---* >>> *-Edwin Quijada >>> *-Developer DataBase >>> *-JQ Microsistemas >>> *-809-849-8087 >>> >>> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo >>> comun" >>> *---* >>> >>> >>> >>> >>> >>> _ >>> Stay up to date on your PC, the Web, and your mobile phone with Windows Live >>> http://clk.atdmt.com/MRT/go/119462413/direct/01/ >>> ___ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> H323 is a dog in Asterisk, JerJer can probaby help you out ot this one. >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> +12024369784 (Skype) >> > > Bye the way, JerJer is the man at NuFone. I don't use H323 but I > believe there are other H323 implementations other than JerJer's you > might want to try. JerJer was never a help to me and purposely ripped > me off for ~$40, this was years ago, I prepaid and the service was > down with no tech support, so I asked for a refund that I never got. > It is in the archives somewhere. > > JerJer/NuFone=bad at least at the beginning. I would look elsewhere. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
On Sat, Jan 31, 2009 at 6:42 PM, Steve Totaro wrote: > On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada > wrote: >> >> >> >> >>> >>> You say "Connected" but do not specify in what fashions you are >>> connecting. That piece of info will be the solution. I have done >>> this many times in many fashions. >>> >> Sorry for my bad english but I dont understand what info you need to know. >> >> You can ask me for anything that u need. >> >> Conection by H323 protocol >> Using The NuFone Network's Open H.323 driver configuration >> Pwlib for asterisk \ >> If u need something else, just ask me.! >> >> >> >> >> >> *---* >> *-Edwin Quijada >> *-Developer DataBase >> *-JQ Microsistemas >> *-809-849-8087 >> >> * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo >> comun" >> *---* >> >> >> >> >> >> _ >> Stay up to date on your PC, the Web, and your mobile phone with Windows Live >> http://clk.atdmt.com/MRT/go/119462413/direct/01/ >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > H323 is a dog in Asterisk, JerJer can probaby help you out ot this one. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > Bye the way, JerJer is the man at NuFone. I don't use H323 but I believe there are other H323 implementations other than JerJer's you might want to try. JerJer was never a help to me and purposely ripped me off for ~$40, this was years ago, I prepaid and the service was down with no tech support, so I asked for a refund that I never got. It is in the archives somewhere. JerJer/NuFone=bad at least at the beginning. I would look elsewhere. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada wrote: > > > > >> >> You say "Connected" but do not specify in what fashions you are >> connecting. That piece of info will be the solution. I have done >> this many times in many fashions. >> > Sorry for my bad english but I dont understand what info you need to know. > > You can ask me for anything that u need. > > Conection by H323 protocol > Using The NuFone Network's Open H.323 driver configuration > Pwlib for asterisk \ > If u need something else, just ask me.! > > > > > > *---* > *-Edwin Quijada > *-Developer DataBase > *-JQ Microsistemas > *-809-849-8087 > > * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo > comun" > *---* > > > > > > _ > Stay up to date on your PC, the Web, and your mobile phone with Windows Live > http://clk.atdmt.com/MRT/go/119462413/direct/01/ > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > H323 is a dog in Asterisk, JerJer can probaby help you out ot this one. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
> > You say "Connected" but do not specify in what fashions you are > connecting. That piece of info will be the solution. I have done > this many times in many fashions. > Sorry for my bad english but I dont understand what info you need to know. You can ask me for anything that u need. Conection by H323 protocol Using The NuFone Network's Open H.323 driver configuration Pwlib for asterisk \ If u need something else, just ask me.! *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun" *---* _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
> You say "Connected" but do not specify in what fashions you are > connecting. That piece of info will be the solution. I have done > this many times in many fashions. > OK. My Avaya is a definity 87000 , Asterisk 1.4.21 *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun" *---* _ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety_112008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Avaya and Asterisk sound one-way
On Sat, Jan 31, 2009 at 5:40 PM, Edwin Quijada wrote: > > > Hi! > I have connected an Avaya System with my asterisk but when I call to avaya > extension I can hear everything but when I speak from Aterisk extension the > person in AVaya cant hear me. > I have seen this issue so much in internet but any solution. > Any help or any cuees?? > > *---* > *-Edwin Quijada > *-Developer DataBase > *-JQ Microsistemas > *-809-849-8087 > > * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo > comun" > *---* > > > > _ > Get 5 GB of storage with Windows Live Hotmail. > http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > You say "Connected" but do not specify in what fashions you are connecting. That piece of info will be the solution. I have done this many times in many fashions. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Avaya and Asterisk sound one-way
Hi! I have connected an Avaya System with my asterisk but when I call to avaya extension I can hear everything but when I speak from Aterisk extension the person in AVaya cant hear me. I have seen this issue so much in internet but any solution. Any help or any cuees?? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * " Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun" *---* _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call without Answer
hi
to read dtmf you need the read app
but some app dont work ok if you dont answer the call first.
David
2009/1/31 msp
> Hi all,
>
> I have a basic understanding about Asterisk and its Dialplan setup.
> I have a sample dialplan scenario to setup as below:
>
> 1. When users registered to asterisk dial any number Asterisk receive the
> call and wait for DTMF entry. (without answering the call)
>
> 2. Based on that DTMF Asterisk route the call to local user or on another
> system.
>
> 3. When the called party answer the call Asterisk does not send Answer back
> to our caller. (means don't send back 200 OK) Until 60 seconds. after that
> Asterisk Send back the answer to caller.
>
> Is this possible to setup such dialplan. How this can be done?
>
> Thanks in Advance
> msp
>
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Re: [asterisk-users] Quiet 24 port POE gig switch
Hello, > I need to put a 24 port Gig PoE switch into a small office – no computer > room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I had a similar problem some days ago. 24-port GBit Switch in the middle of a classroom... I ended up with a kind of "semi-loud" setup: I bought a 3Com 3CBLSG switch (this is without PoE, but there is a PoE version of it). There are two quite noisy 40x40x10 fans inside. I replaced the two with one 40x40x20 ebmPapst silent fan (model 412/2, 18dB), left the other fan offline and mounted the whole thing vertically so that convection supports the remaining fan. I tried with two silent fans (enough space inside), but this still was too noisy. Some measurement indicates the cooling is sufficient this way. But understand that I've no long term data, as I installed this setup just two weeks ago. And of course your warranty is void ;-) Greets, Claus -- CHECON EDV-Consulting und Redaktion Claus Herwig * Barer Straße 70 * 80799 München +49 89 27826981 * Fax 27826982 * c.her...@checon.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata.conf
On Sat, 31 Jan 2009, William Muriithi wrote: > I am however to find zapata.conf. I have searched the whole system for > the name zapata, and none seem to materialize. I am currently interested > to know where it lives. Is it /etc/zapata.conf or > /etc/asterisk/zapata.conf? zapata.conf lives in /etc/asterisk/. (For future reference, "sudo find / | grep -i zapata.conf" may help.) It can be created using genzaptelconf which can be found in zaptel/xpp/utils/. It can be created using "make samples" from your Asterisk source directory, but this may clobber other parts of your configuration. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 107
Sorry, but does that mean if I disable the iptable still I have to do something else? I am talking about Fedora. Also, do I have to give the extern ip to be my DSL router public IP? What it if is dynamic, so I have to put the ddns name? Regards Bilal > > > Ok, I found the problem. I suggested that I disabled > completely my > shorewall-firewall, because there were no rules loaded. But > I were > mistaken... shorewall loads some kernel-modules, especially > ip_nat_sip > and ip_conntrack_sip, and these modules interfere with > asterisk! > > http://www.mail-archive.com/shorewall-us...@lists.sourceforge.net/msg03968.html > > Regards > Holger > > > Holger Latz schrieb: > > Hi all, > > > > I'd like to connect a softphone at home (nat, > dynamic-ip) to a sip-phone > > in the office via asterisk 1.4.21 (nat, fixed-ip). SIP > works well, the > > phone is ringing, but when I pickup the call, > there's no audio on both > > sides. > > > > I debugged the rtp-traffic at home. As long as the > phone is ringing, > > everything is fine. But after the pickup, asterisk > sends a SIP/SDP > > package with its private address (192.168.100.10). > After the softphone > > received this package, it tries to send RTP data to > this address! > > Obviously those packages never reach asterisk... > > > > Does 'externip' just works for SIP and not for > RTP? > > Where does the the internal IP-address come from and > how can I set the > > right one? > > > > > > My configuration: > > > > [general] > > externip = 85.XXX.XXX.XXX > > nat = yes > > localnet = 192.168.100.0/24 > > > > [42] > > deny=0.0.0.0/0.0.0.0 > > disallow=all > > type=friend > > secret=XXX > > qualify=yes > > port=5060 > > pickupgroup= > > permit=0.0.0.0/0.0.0.0 > > nat=yes > > mailbox...@device > > host=dynamic > > dtmfmode=rfc2833 > > dial=SIP/42 > > context=from-internal > > canreinvite=no > > callgroup= > > callerid=device <42> > > allow=alaw > > accountcode= > > call-limit=50 > > > > > > Regards > > Holger ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-users Digest, Vol 54, Issue 107
Sorry but what does the ACL mean and its relation to the bindaddr? Regards Bilal > > 30 jan 2009 kl. 16.59 skrev Mike: > > > hI, > > > > Trying to understand how to setup two PRIs in > sip.conf. Using > > Asterisk 1.4.23. > > > > I have a provider giving me two PRI (different rate > centers) through > > SIP. Both PRI comes in from the same IP on the > provider side, but > > go to two different IPs (both on the same box) on my > side. > > > > How can I setup two different SIP peer, one for each > of the PRIs I > > get, if all I can use to differenciate them are the IP > address?? I > > can't find any obvious setting in the sip.conf > peer settings. The > > general section has "bindaddr" which would > make sense, but since > > it's general and not per peer it's of no use? > > > Interesting question. I don't think you can. The ACLs > only work on > sender's address. Maybe we could consider at some point > implement > local ACLs too. > > /O ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata.conf
William Muriithi wrote: > Hi pals, > > Pardon me if this question sound basic please. This is the first > installation where I have to use the analogue card and therefore a > little lusty. I have googled a lot, but though there is a lot of > information about the above file, none indicate where the file lives. > I have a installed asterisk and zaptel software on a fresh > installation of CentOS 5. This all from source and following closely > the book Asterisk: the future of telephony. > > I am however to find zapata.conf. I have searched the whole system for > the name zapata, and none seem to materialize. I am currently > interested to know where it lives. Is it /etc/zapata.conf or > /etc/asterisk/zapata.conf? It lives in /etc/asterisk/zapata.conf You can find a sample file in the asterisk src 'configs' directory called 'zapata.conf.sample' Andres http://www.neuroredes.com > > Some of the firmware needed also failed to load, or more precisely, > the computer was not in the network, so the firmware was not > downloaded. I have downloaded the said firmware manually, uncompressed > it and placed it in the firmware directory. The kernel still do not > seem to see it. Here is the error I am getting. How does one properly > install firmware for the analogue cards? > > ACPI: PCI Interrupt Link [APC2] enabled at IRQ 17 > ACPI: PCI Interrupt :01:06.0[A] -> Link [APC2] -> GSI 17 (level, > low) -> IRQ 50 > Port 1: Installed -- AUTO FXO (FCC mode) > Port 2: Installed -- AUTO FXO (FCC mode) > Port 3: Installed -- AUTO FXO (FCC mode) > Port 4: Installed -- AUTO FXO (FCC mode) > VPM100: Not Present > VPMADT032: firmware zaptel-fw-vpmadt032.bin not available from userspace > Found a Wildcard TDM: Wildcard TDM410P (4 modules) > lp0: using parport0 (interrupt-driven). > > Regards, > > William > > > >___ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zapata.conf
Hi pals, Pardon me if this question sound basic please. This is the first installation where I have to use the analogue card and therefore a little lusty. I have googled a lot, but though there is a lot of information about the above file, none indicate where the file lives. I have a installed asterisk and zaptel software on a fresh installation of CentOS 5. This all from source and following closely the book Asterisk: the future of telephony. I am however to find zapata.conf. I have searched the whole system for the name zapata, and none seem to materialize. I am currently interested to know where it lives. Is it /etc/zapata.conf or /etc/asterisk/zapata.conf? Some of the firmware needed also failed to load, or more precisely, the computer was not in the network, so the firmware was not downloaded. I have downloaded the said firmware manually, uncompressed it and placed it in the firmware directory. The kernel still do not seem to see it. Here is the error I am getting. How does one properly install firmware for the analogue cards? ACPI: PCI Interrupt Link [APC2] enabled at IRQ 17 ACPI: PCI Interrupt :01:06.0[A] -> Link [APC2] -> GSI 17 (level, low) -> IRQ 50 Port 1: Installed -- AUTO FXO (FCC mode) Port 2: Installed -- AUTO FXO (FCC mode) Port 3: Installed -- AUTO FXO (FCC mode) Port 4: Installed -- AUTO FXO (FCC mode) VPM100: Not Present VPMADT032: firmware zaptel-fw-vpmadt032.bin not available from userspace Found a Wildcard TDM: Wildcard TDM410P (4 modules) lp0: using parport0 (interrupt-driven). Regards, William ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] vmail.cgi - permissions error help
We always install native Asterisk (not when over the other packaged versions) I tried setting the SUID bit on the vmail.cgi file but that didn't help...so I must be missing something. Can someone else suggest a fix? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gondar Monn Sent: January 31, 2009 2:17 AM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Importance: High Have you tried FreePBX ? It allows Asterisk administration via web interface and has module just for that. G. On Fri, Jan 30, 2009 at 7:56 PM, OCG Technical Support wrote: No - the server generates the error: Software error: Hrm, can't seem to open /var/spool/asterisk/voicemail/internalextensions/230/Old/msg.wav For help, please send mail to th From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 10:14 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Importance: High It might be browser security issues? Have you tried with different browsers? On Fri, Jan 30, 2009 at 9:53 PM, OCG Technical Support wrote: Strangely, I can DELETE the messages from the web interface...just playback causes a permission error... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of OCG Technical Support Sent: January 30, 2009 9:40 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? I just tried the vmail.cgi app. Although working, there is clearly a permissions problem preventing playing the wav files. I run Fedora 8, and the patch files (on the wiki) are apparently broken. Does anyone have a solution for fedora? Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Soonthorn Ativanichayaphong Sent: January 30, 2009 7:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] Asterisk VoiceMail: Is there a web interface for checking voicemail? Thank you Lenz. That's exactly what I'm looking for. On Fri, Jan 30, 2009 at 12:21 PM, Lenz Emilitri wrote: We use the default web interface, it comes with a file called vmail.cgi in the defaiult Asterisk tree. Thanks l. 2009/1/30 Soonthorn Ativanichayaphong Hi, I'm very new to Asterisk. I tried VoiceMail() application. I'm able to modify extensions.conf and voicemail.conf to send a voicemail audio file to my email. It works great so far. Now, I'm looking into publishing those voicemail files on a web page. According to http://www.voip-info.org/wiki-Asterisk+VoiceMail, it mentions that Asterisk VoiceMail supports "Web interface for checking of voicemail". Does anyone know where I can find more information about this "Web interface for checking of voicemail" feature? If there is no such thing, what an alternative is? Do you think writing my own AGI script is the right way to go? Is there already an existing AGI scripts or other things that I should install on top of Asterisk that already do something like this? Basically, I try to avoid writing a bunch of codes to retrieve voicemail from my email and build a brand new voicemail web page. If there are something I can reuse, I'd like to use them. Thank you for your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential & Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Soonthorn Ativanichayaphong Software Engineer Yap Inc. -- Confidential & Privileged: This email message is for the sole use of the intended recipient(s) and may contain confidential and privileged info
[asterisk-users] Quiet 24 port POE gig switch
A little off topic but I need to put a 24 port Gig PoE switch into a small office - no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I've researched so far is LOUD... Anyone know of a quiet one? Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] iax clients were unregistered after 30sec
Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language | mailbox | deny | permit | qualify | disallow | allow | ipaddr | port | regseconds | email |date | user_id -+--++--+---+--+--++-+--+-+--+--+-+---+-+--+-+--++-+--+---+-+--+++-+- 9706015 | 9706015 | friend | 5056ed3c | | | no | | | md5 | | | 9706015 | GPHONE | | dynamic | | | || yes | NULL | all | 0.0.0.0 |0 | 0 | pezhman_l...@yahoo.com | 2009-01-31 11:33:10 | 9706015 (1 row) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ideas on how to convert spoken name to text (orwav to text)..speech recognition software?
This would work if you only care that you get very rough phonetic spellings as Don implied. If you think about it humans cannot do any better. I know personally - I have to spell my name all the time. Perhaps your app could ask them to spell their name, which actually has a shot at reliability of not useability. j On Sat, 31 Jan 2009, Kurian Thayil wrote: > Hi Alfred, > > There is a research project by Carnegie Mellon University (CMU) on a very > versatile Speech Recognition Software. Its Sphinx > http://cmusphinx.sourceforge.net/html/cmusphinx.php . This application is in > raw state and the Version 2 of sphinx could be integrated with Asterisk. > Festival (Text to Speech application) that is widely used in asterisk is by > CMU. Refer http://www.voip-info.org/wiki-Sphinx . I hope this gives a pretty > good start. > > Sphinx needs to be trained with a language model. But since your requirement > is just names it should not be complicated. Also have a look at > http://www.speech.cs.cmu.edu/Communicator/ . Something I have not looked > into much (and I don't know if it has anything to do with Asterisk). I hope > this helps. > > Regards, > > Kurian Thayil. > > > On Sat, Jan 31, 2009 at 9:22 AM, Alfred Monticello wrote: > >> >> I wouldn't have a database to compare names to, each one would essentially >> be unique and unknown. It's sounding like this idea may be not >> possible...What high end options are available? I read about lumenvox, but I >> believe that compares to a known list of names (such as a directory, or Yes >> No, Digits, etc) >> >> Hum... >> >> >> -- >> *From:* Don Kelly >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion < >> asterisk-users@lists.digium.com> >> *Sent:* Friday, January 30, 2009 7:41:02 PM >> *Subject:* Re: [asterisk-users] Ideas on how to convert spoken name to >> text (orwav to text)..speech recognition software? >> >> There are solutions ranging from free to many thousands of dollars, with >> effectiveness ranging from nearly worthless to almost pretty good. >> >> >> >> A lot depends on your application. >> >> >> >> The most successful application would match an utterance from a known >> speaker to a known list of a couple dozen names. For example, if I say >> "Alfred Monticello," the application can easily distinguish this from other >> list entries such as "Don Kelly" and "Robert Smith." >> >> >> >> The least successful would attempt to convert an utterance from an unknown >> speaker to text (which is what your inquiry implies). Even if it clearly >> "understands" the speaker, the result could easily be "Alphret Mahntichelo." >> >> --Don >> >> Don Kelly >> PCF Corp >> People Come First >> >> 651 842-1000 >> 888 Don Kell(y) >> 651 842-1001 fax >> -- >> >> *From:* asterisk-users-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Alfred Monticello >> *Sent:* Friday, January 30, 2009 9:25 PM >> *To:* asterisk-users@lists.digium.com >> *Subject:* [asterisk-users] Ideas on how to convert spoken name to text >> (orwav to text)..speech recognition software? >> >> >> >> >> >> I'm interested in taking a persons spoken recorded name (First, Last) and >> converting the two spoken words to text. Is there any solutions out there >> that would make this possible? >> >> >> >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
Jonn Taylor wrote: > Doug Lytle wrote: > >> Jonn Taylor wrote: >> >> >>> Anyone else having problems connecting to >>> http://downloads.digium.com/pub/ ?? >>> >>> >>> >> It would appear it isn't down, but it's not responding to http requests. >> >> Doug >> >> >> >> > Yes, That is the same thing that I am getting. > > Jonn > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Ok, it just started working again. Jonn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
Hi Jonn, in Brazil this http connection to digium pub is OK! working perfectly, see: [DIR] asa/08-Jan-2009 14:40 - [DIR] asterisk/ 29-Jan-2009 17:50 - [DIR] gastman/03-Sep-2008 20:10 - [DIR] gnophone/ 03-Sep-2008 20:30 - [DIR] iaxy/ 03-Sep-2008 20:30 - [DIR] libiax/ 03-Sep-2008 20:11 - [DIR] libpri/ 09-Jan-2009 15:50 - [DIR] polycom/02-Dec-2008 15:30 - [DIR] register/ 03-Sep-2008 20:00 - [DIR] security/ 08-Jan-2009 14:40 - [DIR] support/03-Sep-2008 20:30 - [DIR] telephony/ 23-Sep-2008 11:50 - [DIR] zaptel/ 11-Nov-2008 11:27 - [TXT] README.txt 08-Sep-2008 15:45 192 Best Regards Josué 2009/1/31 Bayardo Sanchez : > i any problem conecct ready > > On Sat, Jan 31, 2009 at 7:37 AM, Jonn Taylor > wrote: >> >> Anyone else having problems connecting to >> http://downloads.digium.com/pub/ ?? >> >> Jonn >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Bayardo Sánchez García > Web Developer - Internet Portals > Linux User: #418392 > Ubuntu User #14171 > America Central - Managua, NI (505) 249-2853 - 4886876 > IM msn messenger: bjsanch...@hotmail.com > Skype: bayardo.sanchez > This email is intended solely for the person or organization to which it is > addressed. It may contain privileged and confidential information. If you > are not the intended recipient, you are prohibited from copying, disclosing > or distributing this email or its contents (as it may be unlawful for you to > do so) or taking any action in reliance on it. If you have received this > email by mistake, please delete it. All e-mail sent to this address will be > received by B.S. Solution e-mail system and is subject to archiving and > review by someone other than the recipient. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
i any problem conecct ready On Sat, Jan 31, 2009 at 7:37 AM, Jonn Taylor wrote: > Anyone else having problems connecting to > http://downloads.digium.com/pub/ ?? > > Jonn > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bayardo Sánchez García Web Developer - Internet Portals Linux User: #418392 Ubuntu User #14171 America Central - Managua, NI (505) 249-2853 - 4886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
Doug Lytle wrote: > Jonn Taylor wrote: > >> Anyone else having problems connecting to >> http://downloads.digium.com/pub/ ?? >> >> > > It would appear it isn't down, but it's not responding to http requests. > > Doug > > > Yes, That is the same thing that I am getting. Jonn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is http://downloads.digium.com/pub/ down???
Jonn Taylor wrote: > Anyone else having problems connecting to > http://downloads.digium.com/pub/ ?? > It would appear it isn't down, but it's not responding to http requests. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is http://downloads.digium.com/pub/ down???
Anyone else having problems connecting to http://downloads.digium.com/pub/ ?? Jonn ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where to find STUN Server howto
Hi Tamer A sip-proxy might be the thing you are searching for... check out siproxd[1] You don't need your own STUN servers to traverse masqueraded firewalls... e.g. you can use stun.ekiga.net and many more... [2] [1] http://siproxd.sourceforge.net/ [2] http://www.stunserver.org/ On Sat, 2009-01-31 at 11:24 +0100, Tamer Higazi wrote: > Hi people! > Do you guys know where to find a STUN Server Howto?! Why?! We all know, > to get Asterisk behind an NAT Router to run, is a bit tricky, and you > might have to fire a lot of holes in your firewall. > > > However, I would appreciate it very much if somebody could give me great > links of how to set up a STUN Server. > > > Tamer > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] where to find STUN Server howto
Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to elegantly set DEVSTATE values after restarting
Yes, I think Danny's hack is the way to go. I'll implement it ASAP. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
