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

[Brad Bergman]

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
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