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

[jihad chalhoub]

swar sir,

can u please unsubscribe me for your list 

b.regards
jihad chalhoub


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> 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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