Hi,
I am new to asterisk , i am getting the following
error,& the /etc/zaptel.conf file entry is as follows
defaultzone=us
loadzone=us
span=1,1,0,esf,b8zs,yellow
bchan=1-23
dchan=24
Parsing '/etc/asterisk/zapata.conf': Found
Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664
zt_open: Unable to specify channel 1: No such device
or address
Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296 mkintf:
Unable to open channel 1: No such device or address
here = 0, tmp->channel = 1, channel = 1
Jul 1 18:33:35 ERROR[16384]: chan_zap.c:7325
setup_zap: Unable to register channel '1-16'
Jul 1 18:33:35 WARNING[16384]: loader.c:312
ast_load_resource: chan_zap.so: load_module failed,
returning -1
== Unregistered channel type 'Tor'
== Unregistered channel type 'Zap'
-- Unregistered channel 1
Jul 1 18:33:35 WARNING[16384]: loader.c:407
load_modules: Loading module chan_zap.so failed!
Regards
Arun
--- asterisk-users-request@lists.digium.com wrote:
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> Today's Topics:
>
> 1. Asterisk ended with exit status 1 (Federico
> Alves)
> 2. Re: Re: teliax [Was: LiveVoip is Bankrupt]
> (Rich Adamson)
> 3. RE: Polycom & VPN trouble (gw@adcomcorp.com)
> 4. Re: Native MoH patch for 1.0.8? (Juan Jose
> Comellas)
> 5. Re: Polycom & VPN trouble (Tim Pushor)
> 6. Re: Re: teliax [Was: LiveVoip is Bankrupt]
> (r00t)
> 7. Newbie Confusion on Call Forward and
> DBput/DBdel (Jeffrey Starin)
> 8. Eicon equipment, BRI Server or PRI?
> (gw@adcomcorp.com)
> 9. Re: Level 3 SIP <--> asterisk (Max Clark)
> 10. Re: Asterisk server with remote monitoring
> capabilities
> (Max Clark)
> 11. How to get the outbound data of agent in queue
> (Gary Li)
> 12. Fw: shoutcast mp3 music onhold with amp
> portal? (hank)
> 13. RE: Re: teliax [Was: LiveVoip is Bankrupt]
> (Jay Milk)
> 14. RE: is teliax down? (Jay Milk)
> 15. RE: LiveVoip is Bankrupt (Jay Milk)
> 16. RE: Fwd: JE TROUVE QUE VOUS N'ETES PAS HONETE!
> (harry gaillac)
> 17. RE: LiveVoip is Bankrupt (Terry H. Gilsenan)
> 18. Can anyone guide me regarding h323.cong ???
> (Adeel -31)
> 19. H323 (Ronald_Wiplinger)
> 20. Re: SixTel? (Erik Espinoza)
> 21. Shoutcast Music On Hold problems? (hank)
> 22. Re: Eicon equipment, BRI Server or PRI? (Armin
> Schindler)
> 23. Re: polycom soundpoint ip 300 (Wilson Pickett)
> 24. RE: RTP session between two end users (Erdem
> HAK?)
> 25. Re: Passing called number in SIP (Andres)
> 26. Re: H323 (Tzafrir Cohen)
> 27. Re: polycom soundpoint ip 300 (harry gaillac)
>
>
>
---------------------------------------------------------------------->
> Message: 1
> Date: Mon, 27 Jun 2005 22:33:54 -0400
> From: "Federico Alves" <sales@minixel.com>
> Subject: [Asterisk-Users] Asterisk ended with exit
> status 1
> To: <asterisk-users@lists.digium.com>
> Message-ID:
> <200506280233.j5S2XvWo008812@ylpvm01.prodigy.net>
> Content-Type: text/plain; charset="us-ascii"
>
> I need some brain-help: I installed the chan_h323
> software, and if I start
> manually Asterisk either by typing safe_asterisk or
> simply asterisk, it
> works, but it fails to start when I insert
> safe_asterisk or simply asterisk
> in /etc/rc.d/rc.local. The asterisk service script
> also fails.
>
> AsteriskAutomatically restarting Asterisk.
> Asterisk ended with exit status 1
> Asterisk died with code 1. Aborting.
> Automatically restarting Asterisk.
> Asterisk ended with exit status 1
> Asterisk died with code 1. Aborting.
> Automatically restarting Asterisk.
> Asterisk ended with exit status 1
> Asterisk died with code 1. Aborting.
> Automatically restarting Asterisk.
> ended with exit status 1
>
> Actually, I paid Jeremy's company to install the
> channel, so I would
> appreciate some effort on his behalf to understand
> the problem, because the
> computer was not showing this behavior before the
> event.
>
>
>
> ------------------------------
>
> Message: 2
> Date: Mon, 27 Jun 2005 21:44:29 -0600
> From: Rich Adamson <radamson@routers.com>
> Subject: Re: [Asterisk-Users] Re: teliax [Was:
> LiveVoip is Bankrupt]
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> <asterisk-users@lists.digium.com>
> Message-ID: <Chameleon.1119927129.adar0@insp8100>
> Content-Type: TEXT/PLAIN; CHARSET=ISO-8859-1
>
>
> > > http://www.nufone.net. I've been using them for
> the past 18 months with zero
> > > technical hassle. Jerjer and Shido6 hang out on
> IRC. Nufone is not a "hand
> > > holding" VOIP provider. You are expected to
> have some clue. This has turned
> > > away a number of people but as I said, they Just
> Work.
> >
> > So I've heard three recommendations for people
> coming from LiveVOIP:
> > nufone, teliax, and voxee. Nufone and teliax both
> are at $0.02/min and
> > voxee is at $0.011/min. No monthly fee plans are
> available from both.
> >
> > I recall hearing of troubles with Nufone support
> awhile back. Have
> > those been resolved?
> >
> > For someone that places outbound calls only, in a
> fairly low volume, is
> > there a recommendation for which one would be best
> for me?
>
> It's probably a $2 decision. Just pick one or two
> and try them.
>
> There are a fair number of people on this list
> (including myself) that
> stay current with multiple itsp's. Every itsp is
> going to have a problem
> now and then, so keeping a couple around isn't a bad
> approach even for
> a home or soho system.
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Mon, 27 Jun 2005 22:58:42 -0400
> From: gw@adcomcorp.com
> Subject: RE: [Asterisk-Users] Polycom & VPN trouble
> To: <asterisk-users@lists.digium.com>
> Message-ID:
>
>
<40E146975809354DB2B9473624174FF011FB98@secure.SOMERS-NY.CENSYS.NET>> Content-Type: text/plain; charset="us-ascii"
>
> Have you considered playing with the timeouts?
>
> Greg
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On
> Behalf Of Tim Pushor
> Sent: Monday, June 27, 2005 4:20 PM
> To: Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: [Asterisk-Users] Polycom & VPN trouble
>
> Hi All,
>
> I am a remote office that is connected to my office
> via openvpn on UDP.
> Voip has always worked well (after discovering
> g729). Initially I used a
> softphone, then an analog set on a sipura 2000, then
> a polycom IP500 (I
> still LOVE this phone). At that point, I started
> noticing that the
> polycom doesn't ring a lot of the time. Since I was
> desperate for a
> phone, I didn't upgrade the firmware, and just got
> the phone going via
> web interface. I was hoping that a firmware update
> and proper
>
=== message truncated ==
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