Hi Lubo, I appreciate your email to help with this issue, but I don't understand your message. I assume your comment about format=slinear is to use format=slinear in phone.conf instead of format=ulaw. If so, how does this get you g723 to iconnect? Using format=g723.1 doesn't seem to work.
Gregg On Wed, 2003-03-12 at 00:50, Lubomir Christov wrote: > Dan, why are you using phonejack with ulaw codec? g723 (format=slinear > only) is working just perfect with phonejack and iconnect :) > > Lubo > > Dan Fernandez wrote: > > I found similar problems. > > > > With my phonejack I can make a call with ulaw with decent quality (I have a > > 64k line). > > > > However, with Messenger I hear a brief horrible noise and thatґs it. > > > > ----- Original Message ----- > > From: "Jim Archer" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Tuesday, March 11, 2003 8:17 PM > > Subject: Re: [Asterisk-Users] iconnect quality? > > > > > > > >>Ok! When I use the 7777 prefix and I allow gsm it does work! And the > >>quality is fine. > >> > >>There are two problems we're having now. > >> > >>1 - From watching the udp fly by, it seems that iconnect does not know > > > > when > > > >>we hang up. For example, if I call a voice mail and it starts giving me > >>its speal and I hang up, iconnect stays connected until the VM hangs up at > >>its end. > >> > >>Next, if we try to call out via iconnect from a sip client extension (like > >>a windows soft phone) all we hear is horrible noise. > >> > >>Has anyone else had these issues? > >> > >>Jim > >> > >> > >>--On Tuesday, March 11, 2003 3:34 PM -0500 Gregg Lebovitz > >><[EMAIL PROTECTED]> wrote: > >> > >> > >>>I haven't play around enough to know whether or not the 7777 prefix is a > >>>toggle. I will do some experimenting and let you know. Right now I am > >>>prefixing all my calls with 7777. > >>> > >>>My experience is that when the carrier's format is G723.1, you can't > >>>hear the incoming voice. When it is in G711 you can. I have made several > >>>calls using G711 and they are acceptable quality. Note that if you > >>>disallow=gsm in the sip.conf file you will get the 488 media errors you > >>>reported earlier. > >>> > >>>Below are my config files for sip and the linejack cards: > >>> > >>>; > >>>; SIP Configuration for Asterisk > >>>; > >>>[general] > >>>port = 5060 ; Port to bind to > >>>bindaddr = 0.0.0.0 ; Address to bind to > >>>context=iconnect ; Default for incoming calls > >>>allow=gsm > >>>allow=ulaw > >>>allow=alaw > >>> > >>>;register=1813342XXXX:[EMAIL PROTECTED] > >>>;register=1202454XXXX:[EMAIL PROTECTED] > >>> > >>>[iconnecthere] > >>>type=friend > >>>username=XXXXXXXX > >>>secret=XXX > >>>host=sipauth.deltathree.com > >>> > >>>; > >>>; Linux Telephony Interface > >>>; > >>>; Configuration file > >>>; > >>>[interfaces] > >>> > >>>mode=dialtone > >>>format=ulaw > >>>echocancel=medium > >>>silencesupression=no > >>> > >>>context=local > >>>context=default > >>> > >>>txgain=100% > >>>rxgain=100% > >>>device => /dev/phone0 > >>> > >>> > >>> > >>>On Tue, 2003-03-11 at 14:28, Jim Archer wrote: > >>> > >>>>Hi Greg and thanks very much... > >>>> > >>>>A few questions... > >>>> > >>>>First, regarding the 7777 prefix, it seemed that this acts as a toggle, > >>>>switching from the one codec to the other. But how do I set which me > >>>>system uses by default? Or does iconnect always use the high bandwidth > >>>>one by default (such that the 7777 always switches to the low > > > > bandwidth > > > >>>>one)? > >>>> > >>>>Next, I am still struggling to understand the SIP options and what goes > >>>>where. Could you please tell me where the format command goes? Is > > > > this > > > >>>>an option on the channel? I thing the allow goes in sip.conf. > >>>> > >>>>Finally, does this have any impact on the problem where the person > >>>>called can not be heard? > >>>> > >>>>Thanks!!! > >>>> > >>>>Jim > >>>> > >>>>--On Tuesday, March 11, 2003 1:35 PM -0500 Gregg Lebovitz > >>>><[EMAIL PROTECTED]> wrote: > >>>> > >>>> > >>>>>Jim, > >>>>> > >>>>>I changed my extensions entry for iconnect to: > >>>>> > >>>>>exten => _1XXXXXXXXXX,1,Dial,SIP/[EMAIL PROTECTED] > >>>>> > >>>>>and my calls work fine with ulaw. I am calling from a linejack card > >>>>>with format=ulaw and SIP with allow=ulaw. > >>>>> > >>>>>Gregg > >>>>> > >>>>>On Mon, 2003-03-10 at 23:01, Jim Archer wrote: > >>>>> > >>>>>>--On Monday, March 10, 2003 4:47 PM -0300 Dan Fernandez > >>>>>><[EMAIL PROTECTED]> wrote: > >>>>>> > >>>>>> > >>>>>>>Iconnect uses codecs g723 and g711 that can be configured for each > >>>>>>>account (you can change them by the 7777 prefix) > >>>>>> > >>>>>>I tried adding the 7777 in front of a number and it reliably > > > > generates > > > >>>>>>error "488 invalid media." > >>>>>> > >>>>>> > >>>>>>_______________________________________________ > >>>>>>Asterisk-Users mailing list > >>>>>>[EMAIL PROTECTED] > >>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>>> > >>>>>_______________________________________________ > >>>>>Asterisk-Users mailing list > >>>>>[EMAIL PROTECTED] > >>>>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>> > >>>> > >>>>_______________________________________________ > >>>>Asterisk-Users mailing list > >>>>[EMAIL PROTECTED] > >>>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >>>_______________________________________________ > >>>Asterisk-Users mailing list > >>>[EMAIL PROTECTED] > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > >> > >>_______________________________________________ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
