[asterisk-users] Call bridged, but no sound

[Norbert Zawodsky]

Hi Brian, hi list,

Brian Candler wrote:
> On Fri, Oct 13, 2006 at 01:35:04AM +0200, Norbert Zawodsky wrote:
>   
>> I've set canreinvite=no on the channel to the SIP provider and it
>> immediately worked. O.k., I'm happy about that but I want to
>> *understand* what's going on here.
>> .
>> My setup is:
>>
>> Asterisk is connected on one side via eth1 to the "outside world" (IP
>> adress 81.223.xxx.xxx) and on the other side via eth0 to the internal
>> LAN (eth0 has IP 192.168.1.200, SNOM phone has 192.168.1.201, ...).
>>     
>
> A good question, for which it's hard to give a short answer :-) ........
>
>   
Thanks for your explainations. Now all that is far more clear to me!
And my *-Box starts working now...

But I have another wierd problem to solve:

I reduced my sip.conf and extension.conf to an absolute minimum.

sip.conf:

[general]
context=from-inode                      ; Default context for incoming calls
realm=zawodsky.at                       ; Realm for digest authentication
defaultexpirey=14400
bindport=5060                           ; UDP Port to bind to (SIP
standard port is 5060)
bindaddr=0.0.0.0                        ; IP address to bind to (0.0.0.0
binds to all)
srvlookup=yes                           ; Enable DNS SRV lookups on
outbound calls
tos=lowdelay                            ;
lowdelay,throughput,reliability,mincost,none
disallow=all                            ; First disallow all codecs
allow=alaw                              ; Allow codecs in order of
preference
allow=ulaw                              ; Allow codecs in order of
preference
allow=gsm                               ; Allow codecs in order of
preference

register => <user>:<passwor>@voip.inode.at:5060

externip = 81.223.241.115               ; Address that we're going to
put in outbound SIP messages
localnet=192.168.1.0/255.255.255.0      ; All RFC 1918 addresses are
local networks
nat=yes                                 ; Global NAT settings  (Affects
all peers and users)

;
; 10 - Chef (Snom360)
;
[10]
type=friend
context=local-clients
host=dynamic
secret=WdCm1g
dtmfmode=rfc2833
callerid=Chef <10>
; callgroup=1
; pickupgroup=1
subscribecontext=local-clients

extensions.conf:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]

[from-inode]
exten => s,1,NoOp(from-inode, EXTEN=${EXTEN})
exten => s,n,Answer()
exten => s,n,Echo()
exten => s,n,Hangup()

[local-clients]

[default]

Now, the behavior I don' understand.

I would assume that all inbound calls should be routed into the 's'
extension. I called * from another phone. The number my SIP provider
gave to me is 8904676, areacode 01. But

  if I call my box dialing my number "8904676", the call is routed to
's'. (I can hear the Echo application talking back to me)

  if I append an extension, regardless of using '10' or any other (fox
example 89046760, 89046761, 890467610, 890467612345), asterisk simply
rejects the call. (The calling phones display says "not possible")

I turned on sip debugging and noted folowing differences in the output
(1st='8904676', 2nd='890467610'):

1st: INVITE sip:s at 81.223.241.115 SIP/2.0
2nd: INVITE sip:01890467610 at 81.223.241.115 SIP/2.0

1st: To: sip:8904676 at p1.voip.inode.at
2nd: To: sip:890467610 at p1.voip.inode.at

1st: From: sip:0132079780 at p1.voip.inode.at;tag=f6554db4ac48a72
2nd: From: sip:0132079780 at p1.voip.inode.at;tag=b9295878fc2630d

1st: Looking for s in from-inode (domain 81.223.241.115)
2nd: Looking for 01890467610 in from-inode (domain 81.223.241.115)

>From this point on, debug output is completely different because 1st
answers and 2nd hangs up.
But why is this so?

Regards Norbert
(And thank you for your patience wiht my beginners questions!)

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