jihad chalhoub
2004-May-24 12:29 UTC
[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@lists.digium.com wrote:> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > 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 > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-admin@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@inaccessnetworks.com> > Organization: inAccess Networks > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Asterisk-oh323 0.6.1 > Compiling problem > Reply-To: asterisk-users@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@basesys.com> > To: asterisk-users@lists.digium.com > Date: Mon, 24 May 2004 12:32:12 -0500 > Reply-To: asterisk-users@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@basesys.com> > > > --__--__-- > > Message: 3 > Date: Mon, 24 May 2004 13:34:02 -0400 (EDT) > From: Steve Creel <screel@turbs.com> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Channelized T1, SIP > phones, HW Echo Canceller > Reply-To: asterisk-users@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 == __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger. http://messenger.yahoo.com/
hank
2004-May-24 13:15 UTC
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs
there is info at the bottum of each and every message that is sent to this list please read info to unsubscribe. hth hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact info: hank@hanksmith.net Email: Same as MSN. ----- Original Message ----- From: "jihad chalhoub" <la_badi@yahoo.com> To: <asterisk-users@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@lists.digium.com wrote: > > Send Asterisk-Users mailing list submissions to > > asterisk-users@lists.digium.com > > > > 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 > > asterisk-users-request@lists.digium.com > > > > You can reach the person managing the list at > > asterisk-users-admin@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@inaccessnetworks.com> > > Organization: inAccess Networks > > To: asterisk-users@lists.digium.com > > Subject: Re: [Asterisk-Users] Asterisk-oh323 0.6.1 > > Compiling problem > > Reply-To: asterisk-users@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@basesys.com> > > To: asterisk-users@lists.digium.com > > Date: Mon, 24 May 2004 12:32:12 -0500 > > Reply-To: asterisk-users@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@basesys.com> > > > > > > --__--__-- > > > > Message: 3 > > Date: Mon, 24 May 2004 13:34:02 -0400 (EDT) > > From: Steve Creel <screel@turbs.com> > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] Channelized T1, SIP > > phones, HW Echo Canceller > > Reply-To: asterisk-users@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 ==> > > > > > __________________________________ > Do you Yahoo!? > Friends. Fun. Try the all-new Yahoo! Messenger. > http://messenger.yahoo.com/ > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users