brad, Just to make sure you understand the settings, not using the 7777 prefix tells iconnect to use uncompressed codecs. Using 7777 sets iconnect into compressed codec mode.
I am experience that same problem as you when I try to use the uncompressed mode. I connect, but cannot hear the other party. Using the 7777 prefix with the gsm codec works. I am using an internet line jack as FXS. My linejack card is configured to use format=ulaw. Also, are you using a NAT/PAT gateway, or are you connected directly to the internet? Gregg On Sun, 2003-03-30 at 05:22, Brad Bergman wrote: > I've tried these settings and I still find that I cannot hear the called > party. I've also tried what feels like every allow/disallow combination > with and without a 7777 prefix and I either get 488 errors, using one > format when the capability is another errors, or completed calls where I > can't hear the called party. > > So pretty much I feel like I'm just going in circles. Any suggestions? > > Brad > > On 20 Mar 2003, Gregg Lebovitz wrote: > > > I remember at some point getting 488 media errors if I didn't enable > > gsm. > > > > Here are my sip.conf and extensions.conf entries. They work for calls > > out to iconnect: > > > > ; > > ; 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 > > disallow=g723.1 > > > > [iconnecthere] > > type=friend > > username=XXXXXXXX > > secret=XXXX > > host=sipauth.deltathree.com > > context=default > > disallow=g723.1 > > allow=gsm > > allow=ulaw > > allow=alaw > > allow=slinear > > > > ;;; extensions.conf > > > > exten => s,1,Wait,1 ; Wait a second, just for fun > > exten => s,2,Answer ; Answer the line > > exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds > > exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds > > exten => s,5,Directory,default > > > > exten => t,1,Goto(#,1) ; If they take too long, give up > > exten => i,1,Playback(invalid) ; "That's not valid, try again" > > > > exten => _1XXXXXXXXXX,1,Dial,SIP/[EMAIL PROTECTED] > > exten => _1XXXXXXXXXX,2,Congestion > > > > > > On Thu, 2003-03-20 at 18:25, Luke Howard wrote: > > > I've found the same. > > > > > > If I make an outgoing call (snom 200 handset), I get about 5 seconds > > > of audio and then it drops out (very occasionally it does work). > > > > > > Incoming calls appear to work, though. > > > > > > -- Executing Goto("SIP/515-Office-143b", "iconnecthere-ulaw|91800XXXXXXX|1") > > > in new stack > > > -- Goto (iconnecthere-ulaw,91800XXXXXXX,1) > > > -- Executing StripMSD("SIP/515-Office-143b", "1") in new stack > > > -- Executing Dial("SIP/515-Office-143b", "SIP/[EMAIL PROTECTED]") in new stack > > > -- Called [EMAIL PROTECTED] > > > -- SIP/iconnecthere-960b answered SIP/515-Office-143b > > > -- Attempting native bridge of SIP/515-Office-143b and SIP/iconnecthere-960b > > > -- Got SIP response 480 "Temporarily not available" back from 213.137.73.178 > > > == Spawn extension (iconnecthere-ulaw, 1800XXXXXXX, 2) exited non-zero on > > > 'SIP/515-Office-143b' > > > > > > SIP config is: > > > > > > [general] > > > port=5060 > > > bindaddr=0.0.0.0 > > > context=sip-remote > > > disallow=all > > > allow=ulaw > > > allow=alaw > > > tos=lowdelay > > > tos=184 > > > register => 1XXXXXXXXXX:[EMAIL PROTECTED] > > > > > > [iconnecthere] > > > type=friend > > > username=XXXXXXXX > > > password=XXXX > > > host=sipauth.deltathree.com > > > context=iconnecthere-ulaw > > > callerid="PADL Software Pty Ltd" <(XXX) XXX XXXX> > > > ;txgain = 5.0; > > > ;rxgain = 5.0; > > > inbanddtmf=1 > > > > > > -- Luke > > > > > > P.S. Is anyone planning on licensing G 723.1 for use with Asterisk? As > > > I understand it, buying a LineJACK won't suffice if the card's DSP is > > > not actually used. > > > -- > > > Luke Howard | PADL Software Pty Ltd | www.padl.com > > > _______________________________________________ > > > 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