[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs

Mon May 24 13:15:15 MST 2004

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----- Original Message -----
From: "jihad chalhoub" <la_badi at yahoo.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, May 24, 2004 12:29 PM
Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs


> swar sir,
>
> can u please unsubscribe me for your list
>
> b.regards
> jihad chalhoub
>
>
> --- asterisk-users-request at lists.digium.com wrote:
> > Send Asterisk-Users mailing list submissions to
> > asterisk-users at lists.digium.com
> >
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> >
> >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > or, via email, send a message with subject or body
> > 'help' to
> > asterisk-users-request at lists.digium.com
> >
> > You can reach the person managing the list at
> > asterisk-users-admin at lists.digium.com
> >
> > When replying, please edit your Subject line so it
> > is more specific
> > than "Re: Contents of Asterisk-Users digest..."
> >
> >
> > Today's Topics:
> >
> >    1. Re: Asterisk-oh323 0.6.1 Compiling problem
> > (Michael Manousos)
> >    2. Re: IP local loop? (Steven Critchfield)
> >    3. Channelized T1, SIP phones, HW Echo Canceller
> > (Steve Creel)
> >    4. Re: Help with IAX , voice Distortion or
> > Breakage. (Alexey Ostrovsky)
> >    5. Re: Where to get 48 volt Power Supplies for
> > Cisco
> >        IP Phones (Greg Boehnlein)
> >    6. extensions/sip from database? (Manuel Wenger)
> >    7. Re: IP local loop? (Shaun Dawson)
> >    8. Re: 2 Sip phones behind un-natted Asterisk
> > (Barry Fawthrop)
> >    9. RE: PRI problem??? (Timothy R. McKee)
> >   10. Re: Where to get 48 volt Power Supplies for
> > Cisco
> >        IP Phones (Nicholas Ruddick)
> >   11. Re: 2 Sip phones behind un-natted Asterisk
> > (Bruce Komito)
> >
> > --__--__--
> >
> > Message: 1
> > Date: Mon, 24 May 2004 20:32:05 +0300
> > From: Michael Manousos
> > <manousos at inaccessnetworks.com>
> > Organization: inAccess Networks
> > To: asterisk-users at lists.digium.com
> > Subject: Re: [Asterisk-Users] Asterisk-oh323 0.6.1
> > Compiling problem
> > Reply-To: asterisk-users at lists.digium.com
> >
> >
> > I need the full output for this (the first lines are
> > missing).
> >
> > Michael.
> >
> > Nicholas Ruddick wrote:
> > > ok done, but now i'm getting different errors -
> > >
> > > /usr/src/pwlib/include/ptlib/args.h:389: virtual
> > outside class declaration
> > > /usr/src/pwlib/include/ptlib/args.h:389:
> > non-member function
> > > `UnknownOption (...)' cannot have `const'
> > > method qualifier
> >
> > [snip...]
> >
> > > in this scope
> > > /usr/src/pwlib/include/ptlib/indchan.h:259:
> > `readChannel' was not
> > > declared in this scope
> > > /usr/src/pwlib/include/ptlib/indchan.h:261:
> > `PChannel' was not declared
> > > in this scope
> > > /usr/src/pwlib/include/ptlib/indchan.h:261:
> > `writeChannel' was not
> > > declared in this scope
> > > /usr/src/pwlib/include/ptlib/indchan.h:263: parse
> > error before `='
> > > /usr/src/pwlib/include/ptlib/indchan.h:265: `BOOL
> > Open (...)' redeclared
> > > as different kind of symbol
> > > /usr/src/pwlib/include/ptlib/indchan.h:229:
> > previous declaration of
> > > `BOOL Open'
> > > /usr/src/pwlib/include/ptlib/indchan.h:229:
> > previous non-function
> > > declaration `BOOL Open'
> > > /usr/src/pwlib/include/ptlib/indchan.h:265:
> > conflicts with function
> > > declaration `BOOL Open (...)'
> > > /usr/src/pwlib/include/ptlib/indchan.h:265:
> > confused by earlier errors,
> > > bailing out
> > > make[1]: *** [asteriskaudio.o] Error 1
> > > make[1]: Leaving directory
> > `/usr/src/asterisk-oh323-0.6.1/wrapper'
> > > make: *** [subdirs_all] Error 1
> > >
> > > Whats this all about, it's still complaining about
> > some audio thing i
> > > just can't work out. I'm using redhat 7.3 btw, i
> > have both the openh323,
> > > pwlib standard, devel and src packages install.
> > Still no joy.
> > >
> > > Thanks,
> > > Nicholas Ruddick
> > >
> > > Pablo Endres wrote:
> > >
> > >> Check your README file again.
> > >>
> > >> In order to compile 0.6.1 you need newer versions
> > of pwlib and
> > >> openh323 (1.6.6 and 1.13.5)
> > >>
> > >> Then it should work just fine
> > >>
> > >> Pablo
> > >>
> > >>
> > >>
> >
> >
> > --__--__--
> >
> > Message: 2
> > Subject: Re: [Asterisk-Users] IP local loop?
> > From: Steven Critchfield <critch at basesys.com>
> > To: asterisk-users at lists.digium.com
> > Date: Mon, 24 May 2004 12:32:12 -0500
> > Reply-To: asterisk-users at lists.digium.com
> >
> > On Mon, 2004-05-24 at 12:19, Shaun Dawson wrote:
> > > Are you guys aware of any providers that do IP
> > local
> > > loop service?  What I want is to get a T-1 from
> > said
> > > provider, plug it into my Cisco router, speak SIP
> > to a
> > > voice gateway upstream, and have phone calls go
> > out
> > > over PSTN from there.
> > >
> > > This is kind of what Vonage and AT&T CallVantage
> > do,
> > > but they are more  geared toward the residential
> > > market, and I want to be able to bring an
> > arbritary
> > > number of lines in.
> >
> > If you want local service, you have to tell us what
> > is local to you,
> > right? Care to finish the details so those on the
> > list can help.
> > --
> > Steven Critchfield  <critch at basesys.com>
> >
> >
> > --__--__--
> >
> > Message: 3
> > Date: Mon, 24 May 2004 13:34:02 -0400 (EDT)
> > From: Steve Creel <screel at turbs.com>
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Channelized T1, SIP
> > phones, HW Echo Canceller
> > Reply-To: asterisk-users at lists.digium.com
> >
> > I have a channelized T1 coming in from our telco,
> > terminated onto a TE405.
> > There are three channelbanks serving internal analog
> > extensions, and about
> > 10 Cisco 7960s.
> >
> > I have no reports of echo on the analog extensions
> > (as expected).  The
> > 7960 users complain of occasional echo (seems like 1
> > in 5 calls).  Only
> > the SIP user hears the echo, not the caller.
> >
> > I have echocancel=yes, echotraining=yes,
> > echocancelwhenbridged=yes.
> > Changes in the taps of echotraining have made things
> > worse, so I have left
> > it alone.
> >
> > I have backed the txgain down, as audio going out on
> > the telco T1 is
> > really hot.  Even at -6dB gain, it is still notably
> > louder from outside
> > than other audio (comparing the ring generated by
> > the telco when calling
> > into asterisk with the ring generated by asterisk
> > calling a station from
> >
> === message truncated ===
>
>
>
>
>
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