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

[Brad Bergman]

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
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > _______________________________________________
> > > Asterisk-Users mailing list
> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
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