[Asterisk-Users] iconnecthere 480 error: is there a workaround?

Sun Mar 30 13:59:13 MST 2003

Brad,

Great. I suspect my difficulty may be related to a NAT/PAT
configuration. The SIP/SDP negotiations go fine, but there could be
problems setting up the incoming RTP session. I wonder how SIP/SDP
decides on what port(s) to listen for the incoming RTP connection.

Gregg

On Sun, 2003-03-30 at 15:18, Brad Bergman wrote:
> I'm not behind a NAT, but of course behind a firewall (duh). I was even
> thinking to myself "this is very much like what happens with IAX when
> there is a firewall issue". So having taken care of that, it works great
> with the same sip.conf settings you have below, and both directions can
> hear each other with the uncompressed codecs used.
> 
> The only problem uncompressed is that I get:
> NOTICE:  File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13 
> received 
> 
> But everything sounds ok. I haven't tried a very lengthy conversation 
> though.
> 
> Brad
> 
> On 30 Mar 2003, Gregg Lebovitz wrote:
> 
> > 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/7777${EXTEN}@iconnecthere
> > > > 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/1800XXXXXXX at iconnecthere") in new stack
> > > > >   -- Called 1800XXXXXXX at iconnecthere
> > > > >   -- 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:XXXX at natrelay.deltathree.com
> > > > > 
> > > > > [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
> > > > > _______________________________________________
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> > > > > Asterisk-Users at lists.digium.com
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> > > > 
> > > 
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