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

Gregg Lebovitz gregg at lebovitz.net
Thu Mar 20 18:44:27 MST 2003


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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